From 0fc0e0e0b89de3becd5f099eae725f13eeecc0d1 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Tue, 25 Jun 2019 11:43:16 +0200 Subject: Remove USAC --- Makefile.am | 9 +- Makefile.vc | 9 +- libAACdec/src/aacdecoder.cpp | 34 +- libAACdec/src/block.cpp | 163 --- libAACdec/src/channel.cpp | 70 +- libAACdec/src/channelinfo.h | 91 -- libAACdec/src/usacdec_ace_d4t64.cpp | 439 -------- libAACdec/src/usacdec_ace_d4t64.h | 117 -- libAACdec/src/usacdec_ace_ltp.cpp | 229 ---- libAACdec/src/usacdec_ace_ltp.h | 128 --- libAACdec/src/usacdec_acelp.cpp | 1296 ---------------------- libAACdec/src/usacdec_acelp.h | 281 ----- libAACdec/src/usacdec_const.h | 203 ---- libAACdec/src/usacdec_fac.cpp | 745 ------------- libAACdec/src/usacdec_fac.h | 191 ---- libAACdec/src/usacdec_lpc.cpp | 1194 --------------------- libAACdec/src/usacdec_lpc.h | 190 ---- libAACdec/src/usacdec_lpd.cpp | 2017 ----------------------------------- libAACdec/src/usacdec_lpd.h | 198 ---- libAACdec/src/usacdec_rom.cpp | 1504 -------------------------- libAACdec/src/usacdec_rom.h | 154 --- 22 files changed, 6 insertions(+), 9258 deletions(-) delete mode 100644 libAACdec/src/usacdec_ace_d4t64.cpp delete mode 100644 libAACdec/src/usacdec_ace_d4t64.h delete mode 100644 libAACdec/src/usacdec_ace_ltp.cpp delete mode 100644 libAACdec/src/usacdec_ace_ltp.h delete mode 100644 libAACdec/src/usacdec_acelp.cpp delete mode 100644 libAACdec/src/usacdec_acelp.h delete mode 100644 libAACdec/src/usacdec_const.h delete mode 100644 libAACdec/src/usacdec_fac.cpp delete mode 100644 libAACdec/src/usacdec_fac.h delete mode 100644 libAACdec/src/usacdec_lpc.cpp delete mode 100644 libAACdec/src/usacdec_lpc.h delete mode 100644 libAACdec/src/usacdec_lpd.cpp delete mode 100644 libAACdec/src/usacdec_lpd.h delete mode 100644 libAACdec/src/usacdec_rom.cpp delete mode 100644 libAACdec/src/usacdec_rom.h diff --git a/Makefile.am b/Makefile.am index 1550d95..d5fd180 100644 --- a/Makefile.am +++ b/Makefile.am @@ -57,14 +57,7 @@ AACDEC_SRC = \ libAACdec/src/channelinfo.cpp \ libAACdec/src/ldfiltbank.cpp \ libAACdec/src/pulsedata.cpp \ - libAACdec/src/stereo.cpp \ - libAACdec/src/usacdec_ace_d4t64.cpp \ - libAACdec/src/usacdec_ace_ltp.cpp \ - libAACdec/src/usacdec_acelp.cpp \ - libAACdec/src/usacdec_fac.cpp \ - libAACdec/src/usacdec_lpc.cpp \ - libAACdec/src/usacdec_lpd.cpp \ - libAACdec/src/usacdec_rom.cpp + libAACdec/src/stereo.cpp AACENC_SRC = \ libAACenc/src/aacEnc_ram.cpp \ diff --git a/Makefile.vc b/Makefile.vc index 54f3744..c236c15 100644 --- a/Makefile.vc +++ b/Makefile.vc @@ -41,14 +41,7 @@ AACDEC_SRC = \ libAACdec/src/channelinfo.cpp \ libAACdec/src/ldfiltbank.cpp \ libAACdec/src/pulsedata.cpp \ - libAACdec/src/stereo.cpp \ - libAACdec/src/usacdec_ace_d4t64.cpp \ - libAACdec/src/usacdec_ace_ltp.cpp \ - libAACdec/src/usacdec_acelp.cpp \ - libAACdec/src/usacdec_fac.cpp \ - libAACdec/src/usacdec_lpc.cpp \ - libAACdec/src/usacdec_lpd.cpp \ - libAACdec/src/usacdec_rom.cpp + libAACdec/src/stereo.cpp AACENC_SRC = \ libAACenc/src/aacEnc_ram.cpp \ diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp index cc3e245..cd57500 100644 --- a/libAACdec/src/aacdecoder.cpp +++ b/libAACdec/src/aacdecoder.cpp @@ -161,8 +161,6 @@ amm-info@iis.fraunhofer.de #include "aacdec_pns.h" -#include "usacdec_lpd.h" - #include "ac_arith_coder.h" #include "tpdec_lib.h" @@ -1942,17 +1940,6 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, goto bail; } } - if (elCh > 0) { - self->pAacDecoderStaticChannelInfo[ch - elCh]->nfRandomSeed = - (ULONG)0x3039; - if (self->elements[el2] == ID_USAC_CPE) { - if (asc->m_sc.m_usacConfig.element[el2].m_stereoConfigIndex != - 1) { - self->pAacDecoderStaticChannelInfo[ch - elCh + 1] - ->nfRandomSeed = (ULONG)0x10932; - } - } - } } /* for each element */ } @@ -2010,11 +1997,6 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc, self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize); - self->pAacDecoderStaticChannelInfo[ch]->last_core_mode = FD_LONG; - self->pAacDecoderStaticChannelInfo[ch]->last_lpd_mode = 255; - - self->pAacDecoderStaticChannelInfo[ch]->last_tcx_pitch = L_DIV; - /* Reset DRC control data for this channel */ aacDecoder_drcInitChannelData( &self->pAacDecoderStaticChannelInfo[ch]->drcData); @@ -2893,10 +2875,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( (self->frameOK && !(flags & AACDEC_CONCEAL)); const int icsIsInvalid = (GetScaleFactorBandsTransmitted(pIcsInfo) > GetScaleFactorBandsTotal(pIcsInfo)); - const int icsInfoUsedinFadeOut = - !(pAacDecoderChannelInfo->renderMode == AACDEC_RENDER_LPD && - pAacDecoderStaticChannelInfo->last_lpd_mode == 0); - if (icsInfoUsedinFadeOut && icsIsInvalid && !mute_release_active) { + if (icsIsInvalid && !mute_release_active) { self->frameOK = 0; } } @@ -2956,16 +2935,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame( 2; } break; case AACDEC_RENDER_LPD: - - CLpd_RenderTimeSignal( - pAacDecoderStaticChannelInfo, pAacDecoderChannelInfo, - pTimeData + offset, self->streamInfo.aacSamplesPerFrame, - &self->samplingRateInfo[streamIndex], - (self->frameOK && !(flags & AACDEC_CONCEAL) && - !frameOk_butConceal), - flags, self->flags[streamIndex]); - - self->extGainDelay = self->streamInfo.aacSamplesPerFrame; + ErrorStatus = AAC_DEC_UNKNOWN; break; default: ErrorStatus = AAC_DEC_UNKNOWN; diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp index a394cd7..b8ca877 100644 --- a/libAACdec/src/block.cpp +++ b/libAACdec/src/block.cpp @@ -107,9 +107,6 @@ amm-info@iis.fraunhofer.de #include "scale.h" #include "FDK_tools_rom.h" -#include "usacdec_fac.h" -#include "usacdec_lpd.h" -#include "usacdec_lpc.h" #include "FDK_trigFcts.h" #include "ac_arith_coder.h" @@ -1010,162 +1007,7 @@ void CBlock_FrequencyToTime( nSpec = 8; break; } - { - int last_frame_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; - - if (pAacDecoderStaticChannelInfo->last_core_mode == LPD) { - INT fac_FB = 1; - if (elFlags & AC_EL_FULLBANDLPD) { - fac_FB = 2; - } - - FIXP_DBL *synth; - - /* Keep some free space at the beginning of the buffer. To be used for - * past data */ - if (!(elFlags & AC_EL_LPDSTEREOIDX)) { - synth = pWorkBuffer1 + ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB); - } else { - synth = pWorkBuffer1 + PIT_MAX_MAX * fac_FB; - } - - int fac_length = - (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) - ? (frameLen >> 4) - : (frameLen >> 3); - - INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; - FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; - - int nbDiv = (elFlags & AC_EL_FULLBANDLPD) ? 2 : 4; - int lFrame = (elFlags & AC_EL_FULLBANDLPD) ? frameLen / 2 : frameLen; - int nbSubfr = - lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ - int LpdSfd = (nbDiv * nbSubfr) >> 1; - int SynSfd = LpdSfd - BPF_SFD; - - FDKmemclear( - pitch, - sizeof( - pitch)); // added to prevent ferret errors in bass_pf_1sf_delay - FDKmemclear(pit_gain, sizeof(pit_gain)); - - /* FAC case */ - if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0 || - pAacDecoderStaticChannelInfo->last_lpd_mode == 4) { - FIXP_DBL fac_buf[LFAC]; - FIXP_LPC *A = pAacDecoderChannelInfo->data.usac.lp_coeff[0]; - - if (!frameOk || last_frame_lost || - (pAacDecoderChannelInfo->data.usac.fac_data[0] == NULL)) { - FDKmemclear(fac_buf, - pAacDecoderChannelInfo->granuleLength * sizeof(FIXP_DBL)); - pAacDecoderChannelInfo->data.usac.fac_data[0] = fac_buf; - pAacDecoderChannelInfo->data.usac.fac_data_e[0] = 0; - } - - INT A_exp; /* linear prediction coefficients exponent */ - { - for (int i = 0; i < M_LP_FILTER_ORDER; i++) { - A[i] = FX_DBL2FX_LPC(fixp_cos( - fMult(pAacDecoderStaticChannelInfo->lpc4_lsf[i], - FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), - LSF_SCALE - LSPARG_SCALE)); - } - - E_LPC_f_lsp_a_conversion(A, A, &A_exp); - } - -#if defined(FDK_ASSERT_ENABLE) - nSamples = -#endif - CLpd_FAC_Acelp2Mdct( - &pAacDecoderStaticChannelInfo->IMdct, synth, - SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), - pAacDecoderChannelInfo->specScale, nSpec, - pAacDecoderChannelInfo->data.usac.fac_data[0], - pAacDecoderChannelInfo->data.usac.fac_data_e[0], fac_length, - frameLen, tl, - FDKgetWindowSlope( - fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - fr, A, A_exp, &pAacDecoderStaticChannelInfo->acelp, - (FIXP_DBL)0, /* FAC gain has already been applied. */ - (last_frame_lost || !frameOk), 1, - pAacDecoderStaticChannelInfo->last_lpd_mode, 0, - pAacDecoderChannelInfo->currAliasingSymmetry); - - } else { -#if defined(FDK_ASSERT_ENABLE) - nSamples = -#endif - imlt_block( - &pAacDecoderStaticChannelInfo->IMdct, synth, - SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), - pAacDecoderChannelInfo->specScale, nSpec, frameLen, tl, - FDKgetWindowSlope( - fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - fl, - FDKgetWindowSlope( - fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - fr, (FIXP_DBL)0, - pAacDecoderChannelInfo->currAliasingSymmetry - ? MLT_FLAG_CURR_ALIAS_SYMMETRY - : 0); - } - FDK_ASSERT(nSamples == frameLen); - - /* The "if" clause is entered both for fullbandLpd mono and - * non-fullbandLpd*. The "else"-> just for fullbandLpd stereo*/ - if (!(elFlags & AC_EL_LPDSTEREOIDX)) { - FDKmemcpy(pitch, pAacDecoderStaticChannelInfo->old_T_pf, - SynSfd * sizeof(INT)); - FDKmemcpy(pit_gain, pAacDecoderStaticChannelInfo->old_gain_pf, - SynSfd * sizeof(FIXP_DBL)); - - for (int i = SynSfd; i < LpdSfd + 3; i++) { - pitch[i] = L_SUBFR; - pit_gain[i] = (FIXP_DBL)0; - } - - if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0) { - pitch[SynSfd] = pitch[SynSfd - 1]; - pit_gain[SynSfd] = pit_gain[SynSfd - 1]; - if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) { - pitch[SynSfd + 1] = pitch[SynSfd]; - pit_gain[SynSfd + 1] = pit_gain[SynSfd]; - } - } - - /* Copy old data to the beginning of the buffer */ - { - FDKmemcpy( - pWorkBuffer1, pAacDecoderStaticChannelInfo->old_synth, - ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB) * sizeof(FIXP_DBL)); - } - - FIXP_DBL *p2_synth = pWorkBuffer1 + (PIT_MAX_MAX * fac_FB); - - /* recalculate pitch gain to allow postfilering on FAC area */ - for (int i = 0; i < SynSfd + 2; i++) { - int T = pitch[i]; - FIXP_DBL gain = pit_gain[i]; - - if (gain > (FIXP_DBL)0) { - gain = get_gain(&p2_synth[i * L_SUBFR * fac_FB], - &p2_synth[(i * L_SUBFR * fac_FB) - fac_FB * T], - L_SUBFR * fac_FB); - pit_gain[i] = gain; - } - } - - bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen, - (LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR, - frameLen - (LpdSfd + 4) * L_SUBFR, outSamples, - pAacDecoderStaticChannelInfo->mem_bpf); - } - - } else /* last_core_mode was not LPD */ { FIXP_DBL *tmp = pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->mdctOutTemp; @@ -1190,11 +1032,6 @@ void CBlock_FrequencyToTime( } FDK_ASSERT(nSamples == frameLen); - - pAacDecoderStaticChannelInfo->last_core_mode = - (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) ? FD_SHORT - : FD_LONG; - pAacDecoderStaticChannelInfo->last_lpd_mode = 255; } #include "ldfiltbank.h" diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp index e17ccf4..fe63756 100644 --- a/libAACdec/src/channel.cpp +++ b/libAACdec/src/channel.cpp @@ -106,9 +106,6 @@ amm-info@iis.fraunhofer.de #include "aacdec_tns.h" #include "FDK_bitstream.h" -#include "usacdec_lpd.h" -#include "usacdec_fac.h" - static void MapMidSideMaskToPnsCorrelation( CAacDecoderChannelInfo *pAacDecoderChannelInfo[2]) { int group; @@ -164,13 +161,6 @@ void CChannelElement_Decode( int maxSfBandsL = 0, maxSfBandsR = 0; int maybe_jstereo = (el_channels > 1); - if (flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA) && el_channels == 2) { - if (pAacDecoderChannelInfo[L]->data.usac.core_mode || - pAacDecoderChannelInfo[R]->data.usac.core_mode) { - maybe_jstereo = 0; - } - } - if (maybe_jstereo) { maxSfBandsL = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[L]->icsInfo); @@ -236,9 +226,6 @@ void CChannelElement_Decode( for (ch = 0; ch < el_channels; ch++) { if (pAacDecoderChannelInfo[ch]->renderMode == AACDEC_RENDER_LPD) { - /* Decode LPD data */ - CLpdChannelStream_Decode(pAacDecoderChannelInfo[ch], - pAacDecoderStaticChannelInfo[ch], flags); } else { UCHAR noSfbs = GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[ch]->icsInfo); @@ -651,13 +638,6 @@ AAC_DECODER_ERROR CChannelElement_Read( break; case core_mode: decision_bit = FDKreadBits(hBs, 1); - pAacDecoderChannelInfo[ch]->data.usac.core_mode = decision_bit; - if ((ch == 1) && (pAacDecoderChannelInfo[0]->data.usac.core_mode != - pAacDecoderChannelInfo[1]->data.usac.core_mode)) { - /* StereoCoreToolInfo(core_mode[ch] ) */ - pAacDecoderChannelInfo[0]->pDynData->RawDataInfo.CommonWindow = 0; - pAacDecoderChannelInfo[1]->pDynData->RawDataInfo.CommonWindow = 0; - } break; case tns_active: pAacDecoderChannelInfo[0]->pDynData->specificTo.usac.tns_active = @@ -671,53 +651,12 @@ AAC_DECODER_ERROR CChannelElement_Read( } break; case lpd_channel_stream: - - { - error = CLpdChannelStream_Read(/* = lpd_channel_stream() */ - hBs, pAacDecoderChannelInfo[ch], - pAacDecoderStaticChannelInfo[ch], - pSamplingRateInfo, flags); - } - pAacDecoderChannelInfo[ch]->renderMode = AACDEC_RENDER_LPD; break; case fac_data: { int fFacDatPresent = FDKreadBit(hBs); - /* Wee need a valid fac_data[0] even if no FAC data is present (as - * temporal buffer) */ - pAacDecoderChannelInfo[ch]->data.usac.fac_data[0] = - pAacDecoderChannelInfo[ch]->data.usac.fac_data0; - - if (fFacDatPresent) { - if (elFlags & AC_EL_LFE) { - error = AAC_DEC_PARSE_ERROR; - break; - } - /* FAC data present, this frame is FD, so the last mode had to be - * ACELP. */ - if (pAacDecoderStaticChannelInfo[ch]->last_core_mode != LPD || - pAacDecoderStaticChannelInfo[ch]->last_lpd_mode != 0) { - pAacDecoderChannelInfo[ch]->data.usac.core_mode_last = LPD; - pAacDecoderChannelInfo[ch]->data.usac.lpd_mode_last = 0; - /* We can't change the past! So look to the future and go ahead! */ - } - CLpd_FAC_Read(hBs, pAacDecoderChannelInfo[ch]->data.usac.fac_data[0], - pAacDecoderChannelInfo[ch]->data.usac.fac_data_e, - CLpd_FAC_getLength( - IsLongBlock(&pAacDecoderChannelInfo[ch]->icsInfo), - pAacDecoderChannelInfo[ch]->granuleLength), - 1, 0); - } else { - if (pAacDecoderStaticChannelInfo[ch]->last_core_mode == LPD && - pAacDecoderStaticChannelInfo[ch]->last_lpd_mode == 0) { - /* ACELP to FD transitons without FAC are possible. That is why we - zero it out (i.e FAC will not be considered in the subsequent - calculations */ - FDKmemclear(pAacDecoderChannelInfo[ch]->data.usac.fac_data0, - LFAC * sizeof(FIXP_DBL)); - } - } + error = AAC_DEC_PARSE_ERROR; } break; case esc2_rvlc: if (flags & AC_ER_RVLC) { @@ -887,13 +826,6 @@ AAC_DECODER_ERROR CChannelElement_Read( if (error != AAC_DEC_OK) { return error; } - - if (elFlags & AC_EL_USAC_NOISE) { - CBlock_ApplyNoise(pAacDecoderChannelInfo[ch], pSamplingRateInfo, - &pAacDecoderStaticChannelInfo[ch]->nfRandomSeed, - band_is_noise); - - } /* if (elFlags & AC_EL_USAC_NOISE) */ } } diff --git a/libAACdec/src/channelinfo.h b/libAACdec/src/channelinfo.h index 04f0012..42c8298 100644 --- a/libAACdec/src/channelinfo.h +++ b/libAACdec/src/channelinfo.h @@ -117,10 +117,6 @@ amm-info@iis.fraunhofer.de #include "aacdec_pns.h" -#include "usacdec_acelp.h" -#include "usacdec_const.h" -#include "usacdec_rom.h" - #include "ac_arith_coder.h" #include "aacdec_drc_types.h" @@ -202,55 +198,6 @@ typedef struct { INT pnsCurrentSeed; - /* LPD memory */ - FIXP_DBL old_synth[PIT_MAX_MAX - L_SUBFR]; - INT old_T_pf[SYN_SFD]; - FIXP_DBL old_gain_pf[SYN_SFD]; - FIXP_DBL mem_bpf[L_FILT + L_SUBFR]; - UCHAR - old_bpf_control_info; /* (1: enable, 0: disable) bpf for past superframe - */ - - USAC_COREMODE last_core_mode; /* core mode used by the decoder in previous - frame. (not signalled by the bitstream, see - CAacDecoderChannelInfo::core_mode_last !! ) - */ - UCHAR last_lpd_mode; /* LPD mode used by the decoder in last LPD subframe - (not signalled by the bitstream, see - CAacDecoderChannelInfo::lpd_mode_last !! ) */ - UCHAR last_last_lpd_mode; /* LPD mode used in second last LPD subframe - (not signalled by the bitstream) */ - UCHAR last_lpc_lost; /* Flag indicating that the previous LPC is lost */ - - FIXP_LPC - lpc4_lsf[M_LP_FILTER_ORDER]; /* Last LPC4 coefficients in LSF domain. */ - FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER]; /* Adaptive mean of LPC - coefficients in LSF domain - for concealment. */ - FIXP_LPC lp_coeff_old[2][M_LP_FILTER_ORDER]; /* Last LPC coefficients in LP - domain. lp_coeff_old[0] is lpc4 (coeffs for - right folding point of last tcx frame), - lp_coeff_old[1] are coeffs for left folding - point of last tcx frame */ - INT lp_coeff_old_exp[2]; - - FIXP_SGL - oldStability; /* LPC coeff stability value from last frame (required for - TCX concealment). */ - UINT numLostLpdFrames; /* Number of consecutive lost subframes. */ - - /* TCX memory */ - FIXP_DBL last_tcx_gain; - INT last_tcx_gain_e; - FIXP_DBL last_alfd_gains[32]; /* Scaled by one bit. */ - SHORT last_tcx_pitch; - UCHAR last_tcx_noise_factor; - - /* ACELP memory */ - CAcelpStaticMem acelp; - - ULONG nfRandomSeed; /* seed value for USAC noise filling random generator */ - CDrcChannelData drcData; CpePersistentData *pCpeStaticData; @@ -299,8 +246,6 @@ typedef shouldBeUnion { * used anymore. */ FIXP_DBL mdctOutTemp[1024]; - FIXP_DBL synth_buf[(PIT_MAX_MAX + SYN_DELAY + L_FRAME_PLUS)]; - FIXP_DBL workBuffer[WB_SECTION_SIZE]; } CWorkBufferCore1; @@ -335,42 +280,6 @@ typedef struct { */ typedef struct { shouldBeUnion { - struct { - FIXP_DBL fac_data0[LFAC]; - SCHAR fac_data_e[4]; - FIXP_DBL - *fac_data[4]; /* Pointers to unused parts of pSpectralCoefficient */ - - UCHAR core_mode; /* current core mode */ - USAC_COREMODE - core_mode_last; /* previous core mode, signalled in the bitstream - (not done by the decoder, see - CAacDecoderStaticChannelInfo::last_core_mode !!)*/ - UCHAR lpd_mode_last; /* previous LPD mode, signalled in the bitstream - (not done by the decoder, see - CAacDecoderStaticChannelInfo::last_core_mode !!)*/ - UCHAR mod[4]; - UCHAR bpf_control_info; /* (1: enable, 0: disable) bpf for current - superframe */ - - FIXP_LPC lsp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction - coefficients in LSP domain */ - FIXP_LPC - lp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction coefficients in - LP domain */ - INT lp_coeff_exp[5]; - FIXP_LPC lsf_adaptive_mean_cand - [M_LP_FILTER_ORDER]; /* concealment: is copied to - CAacDecoderStaticChannelInfo->lsf_adaptive_mean once frame is - assumed to be correct*/ - FIXP_SGL aStability[4]; /* LPC coeff stability values required for ACELP - and TCX (concealment) */ - - CAcelpChannelData acelp[4]; - - FIXP_DBL tcx_gain[4]; - SCHAR tcx_gain_e[4]; - } usac; struct { CPnsData PnsData; /* Not required for USAC */ diff --git a/libAACdec/src/usacdec_ace_d4t64.cpp b/libAACdec/src/usacdec_ace_d4t64.cpp deleted file mode 100644 index 43e06cd..0000000 --- a/libAACdec/src/usacdec_ace_d4t64.cpp +++ /dev/null @@ -1,439 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): - - Description: ACELP - -*******************************************************************************/ - -#include "usacdec_ace_d4t64.h" - -#define L_SUBFR 64 /* Subframe size */ - -/* - * D_ACELP_add_pulse - * - * Parameters: - * pos I: position of pulse - * nb_pulse I: number of pulses - * track I: track - * code O: fixed codebook - * - * Function: - * Add pulses to fixed codebook - * - * Returns: - * void - */ -static void D_ACELP_add_pulse(SHORT pos[], SHORT nb_pulse, SHORT track, - FIXP_COD code[]) { - SHORT i, k; - for (k = 0; k < nb_pulse; k++) { - /* i = ((pos[k] & (16-1))*NB_TRACK) + track; */ - i = ((pos[k] & (16 - 1)) << 2) + track; - if ((pos[k] & 16) == 0) { - code[i] = code[i] + (FIXP_COD)(512 << (COD_BITS - FRACT_BITS)); - } else { - code[i] = code[i] - (FIXP_COD)(512 << (COD_BITS - FRACT_BITS)); - } - } - return; -} -/* - * D_ACELP_decode_1p_N1 - * - * Parameters: - * index I: pulse index - * N I: number of bits for position - * offset I: offset - * pos O: position of the pulse - * - * Function: - * Decode 1 pulse with N+1 bits - * - * Returns: - * void - */ -static void D_ACELP_decode_1p_N1(LONG index, SHORT N, SHORT offset, - SHORT pos[]) { - SHORT pos1; - LONG i, mask; - - mask = ((1 << N) - 1); - /* - * Decode 1 pulse with N+1 bits - */ - pos1 = (SHORT)((index & mask) + offset); - i = ((index >> N) & 1); - if (i == 1) { - pos1 += 16; - } - pos[0] = pos1; - return; -} -/* - * D_ACELP_decode_2p_2N1 - * - * Parameters: - * index I: pulse index - * N I: number of bits for position - * offset I: offset - * pos O: position of the pulse - * - * Function: - * Decode 2 pulses with 2*N+1 bits - * - * Returns: - * void - */ -static void D_ACELP_decode_2p_2N1(LONG index, SHORT N, SHORT offset, - SHORT pos[]) { - SHORT pos1, pos2; - LONG mask, i; - mask = ((1 << N) - 1); - /* - * Decode 2 pulses with 2*N+1 bits - */ - pos1 = (SHORT)(((index >> N) & mask) + offset); - i = (index >> (2 * N)) & 1; - pos2 = (SHORT)((index & mask) + offset); - if ((pos2 - pos1) < 0) { - if (i == 1) { - pos1 += 16; - } else { - pos2 += 16; - } - } else { - if (i == 1) { - pos1 += 16; - pos2 += 16; - } - } - pos[0] = pos1; - pos[1] = pos2; - return; -} -/* - * D_ACELP_decode_3p_3N1 - * - * Parameters: - * index I: pulse index - * N I: number of bits for position - * offset I: offset - * pos O: position of the pulse - * - * Function: - * Decode 3 pulses with 3*N+1 bits - * - * Returns: - * void - */ -static void D_ACELP_decode_3p_3N1(LONG index, SHORT N, SHORT offset, - SHORT pos[]) { - SHORT j; - LONG mask, idx; - - /* - * Decode 3 pulses with 3*N+1 bits - */ - mask = ((1 << ((2 * N) - 1)) - 1); - idx = index & mask; - j = offset; - if (((index >> ((2 * N) - 1)) & 1) == 1) { - j += (1 << (N - 1)); - } - D_ACELP_decode_2p_2N1(idx, N - 1, j, pos); - mask = ((1 << (N + 1)) - 1); - idx = (index >> (2 * N)) & mask; - D_ACELP_decode_1p_N1(idx, N, offset, pos + 2); - return; -} -/* - * D_ACELP_decode_4p_4N1 - * - * Parameters: - * index I: pulse index - * N I: number of bits for position - * offset I: offset - * pos O: position of the pulse - * - * Function: - * Decode 4 pulses with 4*N+1 bits - * - * Returns: - * void - */ -static void D_ACELP_decode_4p_4N1(LONG index, SHORT N, SHORT offset, - SHORT pos[]) { - SHORT j; - LONG mask, idx; - /* - * Decode 4 pulses with 4*N+1 bits - */ - mask = ((1 << ((2 * N) - 1)) - 1); - idx = index & mask; - j = offset; - if (((index >> ((2 * N) - 1)) & 1) == 1) { - j += (1 << (N - 1)); - } - D_ACELP_decode_2p_2N1(idx, N - 1, j, pos); - mask = ((1 << ((2 * N) + 1)) - 1); - idx = (index >> (2 * N)) & mask; - D_ACELP_decode_2p_2N1(idx, N, offset, pos + 2); - return; -} -/* - * D_ACELP_decode_4p_4N - * - * Parameters: - * index I: pulse index - * N I: number of bits for position - * offset I: offset - * pos O: position of the pulse - * - * Function: - * Decode 4 pulses with 4*N bits - * - * Returns: - * void - */ -static void D_ACELP_decode_4p_4N(LONG index, SHORT N, SHORT offset, - SHORT pos[]) { - SHORT j, n_1; - /* - * Decode 4 pulses with 4*N bits - */ - n_1 = N - 1; - j = offset + (1 << n_1); - switch ((index >> ((4 * N) - 2)) & 3) { - case 0: - if (((index >> ((4 * n_1) + 1)) & 1) == 0) { - D_ACELP_decode_4p_4N1(index, n_1, offset, pos); - } else { - D_ACELP_decode_4p_4N1(index, n_1, j, pos); - } - break; - case 1: - D_ACELP_decode_1p_N1((index >> ((3 * n_1) + 1)), n_1, offset, pos); - D_ACELP_decode_3p_3N1(index, n_1, j, pos + 1); - break; - case 2: - D_ACELP_decode_2p_2N1((index >> ((2 * n_1) + 1)), n_1, offset, pos); - D_ACELP_decode_2p_2N1(index, n_1, j, pos + 2); - break; - case 3: - D_ACELP_decode_3p_3N1((index >> (n_1 + 1)), n_1, offset, pos); - D_ACELP_decode_1p_N1(index, n_1, j, pos + 3); - break; - } - return; -} - -/* - * D_ACELP_decode_4t - * - * Parameters: - * index I: index - * mode I: speech mode - * code I: (Q9) algebraic (fixed) codebook excitation - * - * Function: - * 20, 36, 44, 52, 64, 72, 88 bits algebraic codebook. - * 4 tracks x 16 positions per track = 64 samples. - * - * 20 bits 5+5+5+5 --> 4 pulses in a frame of 64 samples. - * 36 bits 9+9+9+9 --> 8 pulses in a frame of 64 samples. - * 44 bits 13+9+13+9 --> 10 pulses in a frame of 64 samples. - * 52 bits 13+13+13+13 --> 12 pulses in a frame of 64 samples. - * 64 bits 2+2+2+2+14+14+14+14 --> 16 pulses in a frame of 64 samples. - * 72 bits 10+2+10+2+10+14+10+14 --> 18 pulses in a frame of 64 samples. - * 88 bits 11+11+11+11+11+11+11+11 --> 24 pulses in a frame of 64 samples. - * - * All pulses can have two (2) possible amplitudes: +1 or -1. - * Each pulse can sixteen (16) possible positions. - * - * codevector length 64 - * number of track 4 - * number of position 16 - * - * Returns: - * void - */ -void D_ACELP_decode_4t64(SHORT index[], int nbits, FIXP_COD code[]) { - LONG L_index; - SHORT k, pos[6]; - - FDKmemclear(code, L_SUBFR * sizeof(FIXP_COD)); - - /* decode the positions and signs of pulses and build the codeword */ - switch (nbits) { - case 12: - for (k = 0; k < 4; k += 2) { - L_index = index[2 * (k / 2) + 1]; - D_ACELP_decode_1p_N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 1, 2 * (index[2 * (k / 2)]) + k / 2, code); - } - break; - case 16: { - int i = 0; - int offset = index[i++]; - offset = (offset == 0) ? 1 : 3; - for (k = 0; k < 4; k++) { - if (k != offset) { - L_index = index[i++]; - D_ACELP_decode_1p_N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 1, k, code); - } - } - } break; - case 20: - for (k = 0; k < 4; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_1p_N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 1, k, code); - } - break; - case 28: - for (k = 0; k < 4 - 2; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 2, k, code); - } - for (k = 2; k < 4; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_1p_N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 1, k, code); - } - break; - case 36: - for (k = 0; k < 4; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 2, k, code); - } - break; - case 44: - for (k = 0; k < 4 - 2; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_3p_3N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 3, k, code); - } - for (k = 2; k < 4; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_2p_2N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 2, k, code); - } - break; - case 52: - for (k = 0; k < 4; k++) { - L_index = (LONG)index[k]; - D_ACELP_decode_3p_3N1(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 3, k, code); - } - break; - case 64: - for (k = 0; k < 4; k++) { - L_index = (((LONG)index[k] << 14) + (LONG)index[k + 4]); - D_ACELP_decode_4p_4N(L_index, 4, 0, pos); - D_ACELP_add_pulse(pos, 4, k, code); - } - break; - default: - FDK_ASSERT(0); - } - return; -} diff --git a/libAACdec/src/usacdec_ace_d4t64.h b/libAACdec/src/usacdec_ace_d4t64.h deleted file mode 100644 index 76bc3d9..0000000 --- a/libAACdec/src/usacdec_ace_d4t64.h +++ /dev/null @@ -1,117 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): - - Description: ACELP - -*******************************************************************************/ - -#ifndef USACDEC_ACE_D4T64_H -#define USACDEC_ACE_D4T64_H - -#include "common_fix.h" - -/* Data type definition for the fixed codebook vector */ -#define FIXP_COD FIXP_SGL -#define FX_COD2FX_DBL(x) (FX_SGL2FX_DBL(x)) -#define FX_DBL2FX_COD(x) FX_DBL2FX_SGL((x) + (FIXP_DBL)0x8000) -#define FX_SGL2FX_COD(x) (x) -#define COD_BITS FRACT_BITS - -void D_ACELP_decode_4t64(SHORT index[], int nbits, FIXP_COD code[]); - -#endif /* USACDEC_ACE_D4T64_H */ diff --git a/libAACdec/src/usacdec_ace_ltp.cpp b/libAACdec/src/usacdec_ace_ltp.cpp deleted file mode 100644 index 5964b49..0000000 --- a/libAACdec/src/usacdec_ace_ltp.cpp +++ /dev/null @@ -1,229 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand - - Description: USAC ACELP LTP filter - -*******************************************************************************/ - -#include "usacdec_ace_ltp.h" - -#include "genericStds.h" -#include "common_fix.h" - -#define UP_SAMP 4 -#define L_INTERPOL2 16 -#define L_SUBFR 64 - -#define A2 FL2FX_SGL(2 * 0.18f) -#define B FL2FX_SGL(0.64f) - -static const LONG Pred_lt4_inter4_2[UP_SAMP][L_INTERPOL2] = { - {(LONG)0x0000FFFC, (LONG)0x0008FFFC, (LONG)0xFFEB004C, (LONG)0xFF50014A, - (LONG)0xFDD90351, (LONG)0xFB2A06CD, (LONG)0xF6920D46, (LONG)0xEBB42B35, - (LONG)0x6D9EEF39, (LONG)0x0618FE0F, (LONG)0xFFE00131, (LONG)0xFE5501C5, - (LONG)0xFE5E015D, (LONG)0xFEF700B6, (LONG)0xFF920037, (LONG)0xFFEC0003}, - {(LONG)0x0002FFF2, (LONG)0x0026FFBD, (LONG)0x005DFF98, (LONG)0x0055FFEF, - (LONG)0xFF89015F, (LONG)0xFD3A04E5, (LONG)0xF7D90DAA, (LONG)0xE67A50EE, - (LONG)0x50EEE67A, (LONG)0x0DAAF7D9, (LONG)0x04E5FD3A, (LONG)0x015FFF89, - (LONG)0xFFEF0055, (LONG)0xFF98005D, (LONG)0xFFBD0026, (LONG)0xFFF20002}, - {(LONG)0x0003FFEC, (LONG)0x0037FF92, (LONG)0x00B6FEF7, (LONG)0x015DFE5E, - (LONG)0x01C5FE55, (LONG)0x0131FFE0, (LONG)0xFE0F0618, (LONG)0xEF396D9E, - (LONG)0x2B35EBB4, (LONG)0x0D46F692, (LONG)0x06CDFB2A, (LONG)0x0351FDD9, - (LONG)0x014AFF50, (LONG)0x004CFFEB, (LONG)0xFFFC0008, (LONG)0xFFFC0000}, - {(LONG)0x0002FFF2, (LONG)0x002BFF9E, (LONG)0x00B9FECE, (LONG)0x01CFFD75, - (LONG)0x035EFBC1, (LONG)0x0521FA0C, (LONG)0x06AAF8C9, (LONG)0x07907852, - (LONG)0x0790F8C9, (LONG)0x06AAFA0C, (LONG)0x0521FBC1, (LONG)0x035EFD75, - (LONG)0x01CFFECE, (LONG)0x00B9FF9E, (LONG)0x002BFFF2, (LONG)0x00020000}}; - -void Pred_lt4(FIXP_DBL exc[], /* in/out: excitation buffer */ - int T0, /* input : integer pitch lag */ - int frac /* input : fraction of lag in range 0..3 */ -) { - int j; - FIXP_DBL *x; - const LONG *interpol; - FIXP_DBL L_sumb, L_sumt; - - x = &exc[-T0 - L_INTERPOL2 + 1]; - - /* remap frac and x: - 0 -> 3 x (unchanged) - 1 -> 0 x-- - 2 -> 1 x-- - 3 -> 2 x-- - */ - - if (--frac < 0) - frac += UP_SAMP; - else - x--; - - j = L_SUBFR + 1; - do { - LONG filt; - FIXP_DBL x0, x1; - FIXP_DBL *xi = x++; - interpol = Pred_lt4_inter4_2[frac]; - int i = 3; - - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultDiv2(x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultDiv2(x1, (FIXP_SGL)((SHORT)filt)); - do { - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); - - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); - - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); - - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); - - filt = *interpol++; - x0 = *xi++; - x1 = *xi++; - L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16))); - L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt)); - } while (--i != 0); - - L_sumb <<= 1; - L_sumb = fAddSaturate(L_sumt << 1, L_sumb); - *exc++ = L_sumb; - } while (--j != 0); - return; -} - -void Pred_lt4_postfilter(FIXP_DBL exc[] /* in/out: excitation buffer */ -) { - /* - exc[i] = A*exc[i-1] + B*exc[i] + A*exc[i+1] - exc[i+1] = A*exc[i] + B*exc[i+1] + A*exc[i+2] ; i = 0:2:62 - */ - int i; - FIXP_DBL sum0, sum1, a_exc0, a_exc1; - a_exc0 = fMultDiv2(A2, exc[-1]); - a_exc1 = fMultDiv2(A2, exc[0]); - - /* ARM926: 22 cycles/iteration */ - for (i = 0; i < L_SUBFR; i += 2) { - sum0 = a_exc0 + fMult(B, exc[i]); - sum1 = a_exc1 + fMult(B, exc[i + 1]); - a_exc0 = fMultDiv2(A2, exc[i + 1]); - a_exc1 = fMultDiv2(A2, exc[i + 2]); - exc[i] = sum0 + a_exc0; - exc[i + 1] = sum1 + a_exc1; - } - return; -} diff --git a/libAACdec/src/usacdec_ace_ltp.h b/libAACdec/src/usacdec_ace_ltp.h deleted file mode 100644 index 5128acd..0000000 --- a/libAACdec/src/usacdec_ace_ltp.h +++ /dev/null @@ -1,128 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand - - Description: USAC ACELP LTP filter - -*******************************************************************************/ - -#ifndef USACDEC_ACE_LTP_H -#define USACDEC_ACE_LTP_H - -#include "common_fix.h" - -/** - * \brief Compute the initial adaptive codebook excitation v'(n) by - * interpolating the past excitation vector u'(n). - * \param exc points to adaptive codebook of current subframe (input/output) - * \param T0 integer part of decoded pitch lag (input) - * \param frac fractional part of decoded pitch lag (0..3) (input) - */ -void Pred_lt4(FIXP_DBL exc[], /* in/out: excitation buffer */ - int T0, /* input : integer pitch lag */ - int frac /* input : fraction of lag */ -); - -/** - * \brief Compute the adaptive codebook excitation v(n) in case of - * ltp_filtering_flag == 0. - * \param exc points to adaptive codebook of current subframe (input/output) - */ -void Pred_lt4_postfilter(FIXP_DBL exc[] /* in/out: excitation buffer */ -); - -#endif /* USACDEC_ACE_LTP_H */ diff --git a/libAACdec/src/usacdec_acelp.cpp b/libAACdec/src/usacdec_acelp.cpp deleted file mode 100644 index a606459..0000000 --- a/libAACdec/src/usacdec_acelp.cpp +++ /dev/null @@ -1,1296 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand - - Description: USAC ACELP frame decoder - -*******************************************************************************/ - -#include "usacdec_acelp.h" - -#include "usacdec_ace_d4t64.h" -#include "usacdec_ace_ltp.h" -#include "usacdec_rom.h" -#include "usacdec_lpc.h" -#include "genericStds.h" - -#define PIT_FR2_12k8 128 /* Minimum pitch lag with resolution 1/2 */ -#define PIT_FR1_12k8 160 /* Minimum pitch lag with resolution 1 */ -#define TILT_CODE2 \ - FL2FXCONST_SGL(0.3f * 2.0f) /* ACELP code pre-emphasis factor ( *2 ) */ -#define PIT_SHARP \ - FL2FXCONST_SGL(0.85f) /* pitch sharpening factor */ -#define PREEMPH_FAC \ - FL2FXCONST_SGL(0.68f) /* ACELP synth pre-emphasis factor */ - -#define ACELP_HEADROOM 1 -#define ACELP_OUTSCALE (MDCT_OUT_HEADROOM - ACELP_HEADROOM) - -/** - * \brief Calculate pre-emphasis (1 - mu z^-1) on input signal. - * \param[in] in pointer to input signal; in[-1] is also needed. - * \param[out] out pointer to output signal. - * \param[in] L length of filtering. - */ -/* static */ -void E_UTIL_preemph(const FIXP_DBL *in, FIXP_DBL *out, INT L) { - int i; - - for (i = 0; i < L; i++) { - out[i] = in[i] - fMult(PREEMPH_FAC, in[i - 1]); - } - - return; -} - -/** - * \brief Calculate de-emphasis 1/(1 - TILT_CODE z^-1) on innovative codebook - * vector. - * \param[in,out] x innovative codebook vector. - */ -static void Preemph_code( - FIXP_COD x[] /* (i/o) : input signal overwritten by the output */ -) { - int i; - FIXP_DBL L_tmp; - - /* ARM926: 12 cycles per sample */ - for (i = L_SUBFR - 1; i > 0; i--) { - L_tmp = FX_COD2FX_DBL(x[i]); - L_tmp -= fMultDiv2(x[i - 1], TILT_CODE2); - x[i] = FX_DBL2FX_COD(L_tmp); - } -} - -/** - * \brief Apply pitch sharpener to the innovative codebook vector. - * \param[in,out] x innovative codebook vector. - * \param[in] pit_lag decoded pitch lag. - */ -static void Pit_shrp( - FIXP_COD x[], /* in/out: impulse response (or algebraic code) */ - int pit_lag /* input : pitch lag */ -) { - int i; - FIXP_DBL L_tmp; - - for (i = pit_lag; i < L_SUBFR; i++) { - L_tmp = FX_COD2FX_DBL(x[i]); - L_tmp += fMult(x[i - pit_lag], PIT_SHARP); - x[i] = FX_DBL2FX_COD(L_tmp); - } - - return; -} - - /** - * \brief Calculate Quantized codebook gain, Quantized pitch gain and unbiased - * Innovative code vector energy. - * \param[in] index index of quantizer. - * \param[in] code innovative code vector with exponent = SF_CODE. - * \param[out] gain_pit Quantized pitch gain g_p with exponent = SF_GAIN_P. - * \param[out] gain_code Quantized codebook gain g_c. - * \param[in] mean_ener mean_ener defined in open-loop (2 bits), exponent = 7. - * \param[out] E_code unbiased innovative code vector energy. - * \param[out] E_code_e exponent of unbiased innovative code vector energy. - */ - -#define SF_MEAN_ENER_LG10 9 - -/* pow(10.0, {18, 30, 42, 54}/20.0) /(float)(1<>= 1; - } else { - ener_code_e -= 6; - } - gcode_inov = invSqrtNorm2(ener_code, &gcode0_e); - gcode_inov_e = gcode0_e - (ener_code_e >> 1); - - if (bfi) { - FIXP_DBL tgcode; - FIXP_SGL tgpit; - - tgpit = *past_gpit; - - if (tgpit > FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P))) { - tgpit = FL2FXCONST_SGL(0.95f / (1 << SF_GAIN_P)); - } else if (tgpit < FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P))) { - tgpit = FL2FXCONST_SGL(0.5f / (1 << SF_GAIN_P)); - } - *gain_pit = tgpit; - tgpit = FX_DBL2FX_SGL(fMult(tgpit, FL2FXCONST_DBL(0.95f))); - *past_gpit = tgpit; - - tgpit = FL2FXCONST_SGL(1.4f / (1 << SF_GAIN_P)) - tgpit; - tgcode = fMult(*past_gcode, tgpit) << SF_GAIN_P; - *gain_code = scaleValue(fMult(tgcode, gcode_inov), gcode_inov_e); - *past_gcode = tgcode; - - return; - } - - /*-------------- Decode gains ---------------*/ - /* - gcode0 = pow(10.0, (float)mean_ener/20.0); - gcode0 = gcode0 / sqrt(ener_code/L_SUBFR); - */ - gcode0 = pow_10_mean_energy[mean_ener_bits]; - gcode0 = fMultDiv2(gcode0, gcode_inov); - gcode0_e = gcode0_e + SF_MEAN_ENER_LG10 - (ener_code_e >> 1) + 1; - - i = index << 1; - *gain_pit = fdk_t_qua_gain7b[i]; /* adaptive codebook gain */ - /* t_qua_gain[ind2p1] : fixed codebook gain correction factor */ - Ltmp = fMult(fdk_t_qua_gain7b[i + 1], gcode0); - *gain_code = scaleValue(Ltmp, gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B); - - /* update bad frame handler */ - *past_gpit = *gain_pit; - - /*-------------------------------------------------------- - past_gcode = gain_code/gcode_inov - --------------------------------------------------------*/ - { - FIXP_DBL gcode_m; - INT gcode_e; - - gcode_m = fDivNormHighPrec(Ltmp, gcode_inov, &gcode_e); - gcode_e += (gcode0_e - SF_GAIN_C + SF_QUA_GAIN7B) - (gcode_inov_e); - *past_gcode = scaleValue(gcode_m, gcode_e); - } -} - -/** - * \brief Calculate period/voicing factor r_v - * \param[in] exc pitch excitation. - * \param[in] gain_pit gain of pitch g_p. - * \param[in] gain_code gain of code g_c. - * \param[in] gain_code_e exponent of gain of code. - * \param[in] ener_code unbiased innovative code vector energy. - * \param[in] ener_code_e exponent of unbiased innovative code vector energy. - * \return period/voice factor r_v (-1=unvoiced to 1=voiced), exponent SF_PFAC. - */ -static FIXP_DBL calc_period_factor(FIXP_DBL exc[], FIXP_SGL gain_pit, - FIXP_DBL gain_code, FIXP_DBL ener_code, - int ener_code_e) { - int ener_exc_e, L_tmp_e, s = 0; - FIXP_DBL ener_exc, L_tmp; - FIXP_DBL period_fac; - - /* energy of pitch excitation */ - ener_exc = (FIXP_DBL)0; - for (int i = 0; i < L_SUBFR; i++) { - ener_exc += fPow2Div2(exc[i]) >> s; - if (ener_exc >= FL2FXCONST_DBL(0.5f)) { - ener_exc >>= 1; - s++; - } - } - - ener_exc_e = fNorm(ener_exc); - ener_exc = fMult(ener_exc << ener_exc_e, fPow2(gain_pit)); - if (ener_exc != (FIXP_DBL)0) { - ener_exc_e = 2 * SF_EXC + 1 + 2 * SF_GAIN_P - ener_exc_e + s; - } else { - ener_exc_e = 0; - } - - /* energy of innovative code excitation */ - /* L_tmp = ener_code * gain_code*gain_code; */ - L_tmp_e = fNorm(gain_code); - L_tmp = fPow2(gain_code << L_tmp_e); - L_tmp = fMult(ener_code, L_tmp); - L_tmp_e = 2 * SF_GAIN_C + ener_code_e - 2 * L_tmp_e; - - /* Find common exponent */ - { - FIXP_DBL num, den; - int exp_diff; - - exp_diff = ener_exc_e - L_tmp_e; - if (exp_diff >= 0) { - ener_exc >>= 1; - if (exp_diff <= DFRACT_BITS - 2) { - L_tmp >>= exp_diff + 1; - } else { - L_tmp = (FIXP_DBL)0; - } - den = ener_exc + L_tmp; - if (ener_exc_e < DFRACT_BITS - 1) { - den += scaleValue(FL2FXCONST_DBL(0.01f), -ener_exc_e - 1); - } - } else { - if (exp_diff >= -(DFRACT_BITS - 2)) { - ener_exc >>= 1 - exp_diff; - } else { - ener_exc = (FIXP_DBL)0; - } - L_tmp >>= 1; - den = ener_exc + L_tmp; - if (L_tmp_e < DFRACT_BITS - 1) { - den += scaleValue(FL2FXCONST_DBL(0.01f), -L_tmp_e - 1); - } - } - num = (ener_exc - L_tmp); - num >>= SF_PFAC; - - if (den > (FIXP_DBL)0) { - if (ener_exc > L_tmp) { - period_fac = schur_div(num, den, 16); - } else { - period_fac = -schur_div(-num, den, 16); - } - } else { - period_fac = (FIXP_DBL)MAXVAL_DBL; - } - } - - /* exponent = SF_PFAC */ - return period_fac; -} - -/*------------------------------------------------------------* - * noise enhancer * - * ~~~~~~~~~~~~~~ * - * - Enhance excitation on noise. (modify gain of code) * - * If signal is noisy and LPC filter is stable, move gain * - * of code 1.5 dB toward gain of code threshold. * - * This decrease by 3 dB noise energy variation. * - *------------------------------------------------------------*/ -/** - * \brief Enhance excitation on noise. (modify gain of code) - * \param[in] gain_code Quantized codebook gain g_c, exponent = SF_GAIN_C. - * \param[in] period_fac periodicity factor, exponent = SF_PFAC. - * \param[in] stab_fac stability factor, exponent = SF_STAB. - * \param[in,out] p_gc_threshold modified gain of previous subframe. - * \return gain_code smoothed gain of code g_sc, exponent = SF_GAIN_C. - */ -static FIXP_DBL -noise_enhancer(/* (o) : smoothed gain g_sc SF_GAIN_C */ - FIXP_DBL gain_code, /* (i) : Quantized codebook gain SF_GAIN_C */ - FIXP_DBL period_fac, /* (i) : periodicity factor (-1=unvoiced to - 1=voiced), SF_PFAC */ - FIXP_SGL stab_fac, /* (i) : stability factor (0 <= ... < 1.0) - SF_STAB */ - FIXP_DBL - *p_gc_threshold) /* (io): gain of code threshold SF_GAIN_C */ -{ - FIXP_DBL fac, L_tmp, gc_thres; - - gc_thres = *p_gc_threshold; - - L_tmp = gain_code; - if (L_tmp < gc_thres) { - L_tmp += fMultDiv2(gain_code, - FL2FXCONST_SGL(2.0 * 0.19f)); /* +1.5dB => *(1.0+0.19) */ - if (L_tmp > gc_thres) { - L_tmp = gc_thres; - } - } else { - L_tmp = fMult(gain_code, - FL2FXCONST_SGL(1.0f / 1.19f)); /* -1.5dB => *10^(-1.5/20) */ - if (L_tmp < gc_thres) { - L_tmp = gc_thres; - } - } - *p_gc_threshold = L_tmp; - - /* voicing factor lambda = 0.5*(1-period_fac) */ - /* gain smoothing factor S_m = lambda*stab_fac (=fac) - = 0.5(stab_fac - stab_fac * period_fac) */ - fac = (FX_SGL2FX_DBL(stab_fac) >> (SF_PFAC + 1)) - - fMultDiv2(stab_fac, period_fac); - /* fac_e = SF_PFAC + SF_STAB */ - FDK_ASSERT(fac >= (FIXP_DBL)0); - - /* gain_code = (float)((fac*tmp) + ((1.0-fac)*gain_code)); */ - gain_code = fMult(fac, L_tmp) - - fMult(FL2FXCONST_DBL(-1.0f / (1 << (SF_PFAC + SF_STAB))) + fac, - gain_code); - gain_code <<= (SF_PFAC + SF_STAB); - - return gain_code; -} - -/** - * \brief Update adaptive codebook u'(n) (exc) - * Enhance pitch of c(n) and build post-processed excitation u(n) (exc2) - * \param[in] code innovative codevector c(n), exponent = SF_CODE. - * \param[in,out] exc filtered adaptive codebook v(n), exponent = SF_EXC. - * \param[in] gain_pit adaptive codebook gain, exponent = SF_GAIN_P. - * \param[in] gain_code innovative codebook gain g_c, exponent = SF_GAIN_C. - * \param[in] gain_code_smoothed smoothed innov. codebook gain g_sc, exponent = - * SF_GAIN_C. - * \param[in] period_fac periodicity factor r_v, exponent = SF_PFAC. - * \param[out] exc2 post-processed excitation u(n), exponent = SF_EXC. - */ -void BuildAdaptiveExcitation( - FIXP_COD code[], /* (i) : algebraic codevector c(n) Q9 */ - FIXP_DBL exc[], /* (io): filtered adaptive codebook v(n) Q15 */ - FIXP_SGL gain_pit, /* (i) : adaptive codebook gain g_p Q14 */ - FIXP_DBL gain_code, /* (i) : innovative codebook gain g_c Q16 */ - FIXP_DBL gain_code_smoothed, /* (i) : smoothed innov. codebook gain g_sc - Q16 */ - FIXP_DBL period_fac, /* (i) : periodicity factor r_v Q15 */ - FIXP_DBL exc2[] /* (o) : post-processed excitation u(n) Q15 */ -) { -/* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory! - If exc2[i] is written, code[i] will be destroyed! -*/ -#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC) - - int i; - FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth; - - FIXP_COD code_i; - FIXP_DBL cpe_code_smooth, cpe_code_smooth_prev; - - /* cpe = (1+r_v)/8 * 2 ; ( SF = -1) */ - cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f); - - /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); /* v(0)*g_p */ - *exc++ = tmp + (fMultDiv2(code[0], gain_code) << SF); - - /* u(n) */ - code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed) - << SF; /* c(0) * g_sc */ - code_i = *code++; - code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */ - tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */ - cpe_code_smooth = fMultDiv2(cpe, code_smooth); - *exc2++ = tmp - cpe_code_smooth; - cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev); - - i = L_SUBFR - 2; - do /* ARM926: 22 cycles per iteration */ - { - /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); - *exc++ = tmp + (fMultDiv2(code_i, gain_code) << SF); - /* u(n) */ - tmp += code_smooth; /* += g_sc * c(i) */ - tmp -= cpe_code_smooth_prev; - cpe_code_smooth_prev = cpe_code_smooth; - code_i = *code++; - code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; - cpe_code_smooth = fMultDiv2(cpe, code_smooth); - *exc2++ = tmp - cpe_code_smooth; /* tmp - c_pe * g_sc * c(i+1) */ - } while (--i != 0); - - /* u'(n) */ - tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); - *exc = tmp + (fMultDiv2(code_i, gain_code) << SF); - /* u(n) */ - tmp += code_smooth; - tmp -= cpe_code_smooth_prev; - *exc2++ = tmp; - - return; -} - -/** - * \brief Interpolate LPC vector in LSP domain for current subframe and convert - * to LP domain - * \param[in] lsp_old LPC vector (LSP domain) corresponding to the beginning of - * current ACELP frame. - * \param[in] lsp_new LPC vector (LSP domain) corresponding to the end of - * current ACELP frame. - * \param[in] subfr_nr number of current ACELP subframe 0..3. - * \param[in] nb_subfr total number of ACELP subframes in this frame. - * \param[out] A LP filter coefficients for current ACELP subframe, exponent = - * SF_A_COEFFS. - */ -/* static */ -void int_lpc_acelp( - const FIXP_LPC lsp_old[], /* input : LSPs from past frame */ - const FIXP_LPC lsp_new[], /* input : LSPs from present frame */ - int subfr_nr, int nb_subfr, - FIXP_LPC - A[], /* output: interpolated LP coefficients for current subframe */ - INT *A_exp) { - int i; - FIXP_LPC lsp_interpol[M_LP_FILTER_ORDER]; - FIXP_SGL fac_old, fac_new; - - FDK_ASSERT((nb_subfr == 3) || (nb_subfr == 4)); - - fac_old = lsp_interpol_factor[nb_subfr & 0x1][(nb_subfr - 1) - subfr_nr]; - fac_new = lsp_interpol_factor[nb_subfr & 0x1][subfr_nr]; - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp_interpol[i] = FX_DBL2FX_LPC( - (fMultDiv2(lsp_old[i], fac_old) + fMultDiv2(lsp_new[i], fac_new)) << 1); - } - - E_LPC_f_lsp_a_conversion(lsp_interpol, A, A_exp); - - return; -} - -/** - * \brief Perform LP synthesis by filtering the post-processed excitation u(n) - * through the LP synthesis filter 1/A(z) - * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS. - * \param[in] length length of input/output signal. - * \param[in] x post-processed excitation u(n). - * \param[in,out] y LP synthesis signal and filter memory - * y[-M_LP_FILTER_ORDER..-1]. - */ - -/* static */ -void Syn_filt(const FIXP_LPC a[], /* (i) : a[m] prediction coefficients Q12 */ - const INT a_exp, - INT length, /* (i) : length of input/output signal (64|128) */ - FIXP_DBL x[], /* (i) : input signal Qx */ - FIXP_DBL y[] /* (i/o) : filter states / output signal Qx-s*/ -) { - int i, j; - FIXP_DBL L_tmp; - - for (i = 0; i < length; i++) { - L_tmp = (FIXP_DBL)0; - - for (j = 0; j < M_LP_FILTER_ORDER; j++) { - L_tmp -= fMultDiv2(a[j], y[i - (j + 1)]) >> (LP_FILTER_SCALE - 1); - } - - L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE); - y[i] = fAddSaturate(L_tmp, x[i]); - } - - return; -} - -/** - * \brief Calculate de-emphasis 1/(1 - mu z^-1) on input signal. - * \param[in] x input signal. - * \param[out] y output signal. - * \param[in] L length of signal. - * \param[in,out] mem memory (signal[-1]). - */ -/* static */ -void Deemph(FIXP_DBL *x, FIXP_DBL *y, int L, FIXP_DBL *mem) { - int i; - FIXP_DBL yi = *mem; - - for (i = 0; i < L; i++) { - FIXP_DBL xi = x[i] >> 1; - xi = fMultAddDiv2(xi, PREEMPH_FAC, yi); - yi = SATURATE_LEFT_SHIFT(xi, 1, 32); - y[i] = yi; - } - *mem = yi; - return; -} - -/** - * \brief Compute the LP residual by filtering the input speech through the - * analysis filter A(z). - * \param[in] a LP filter coefficients, exponent = SF_A_COEFFS - * \param[in] x input signal (note that values x[-m..-1] are needed), exponent = - * SF_SYNTH - * \param[out] y output signal (residual), exponent = SF_EXC - * \param[in] l length of filtering - */ -/* static */ -void E_UTIL_residu(const FIXP_LPC *a, const INT a_exp, FIXP_DBL *x, FIXP_DBL *y, - INT l) { - FIXP_DBL s; - INT i, j; - - /* (note that values x[-m..-1] are needed) */ - for (i = 0; i < l; i++) { - s = (FIXP_DBL)0; - - for (j = 0; j < M_LP_FILTER_ORDER; j++) { - s += fMultDiv2(a[j], x[i - j - 1]) >> (LP_FILTER_SCALE - 1); - } - - s = scaleValue(s, a_exp + LP_FILTER_SCALE); - y[i] = fAddSaturate(s, x[i]); - } - - return; -} - -/* use to map subfr number to number of bits used for acb_index */ -static const UCHAR num_acb_idx_bits_table[2][NB_SUBFR] = { - {9, 6, 9, 6}, /* coreCoderFrameLength == 1024 */ - {9, 6, 6, 0} /* coreCoderFrameLength == 768 */ -}; - -static int DecodePitchLag(HANDLE_FDK_BITSTREAM hBs, - const UCHAR num_acb_idx_bits, - const int PIT_MIN, /* TMIN */ - const int PIT_FR2, /* TFR2 */ - const int PIT_FR1, /* TFR1 */ - const int PIT_MAX, /* TMAX */ - int *pT0, int *pT0_frac, int *pT0_min, int *pT0_max) { - int acb_idx; - int error = 0; - int T0, T0_frac; - - FDK_ASSERT((num_acb_idx_bits == 9) || (num_acb_idx_bits == 6)); - - acb_idx = FDKreadBits(hBs, num_acb_idx_bits); - - if (num_acb_idx_bits == 6) { - /* When the pitch value is encoded on 6 bits, a pitch resolution of 1/4 is - always used in the range [T1-8, T1+7.75], where T1 is nearest integer to - the fractional pitch lag of the previous subframe. - */ - T0 = *pT0_min + acb_idx / 4; - T0_frac = acb_idx & 0x3; - } else { /* num_acb_idx_bits == 9 */ - /* When the pitch value is encoded on 9 bits, a fractional pitch delay is - used with resolutions 0.25 in the range [TMIN, TFR2-0.25], resolutions - 0.5 in the range [TFR2, TFR1-0.5], and integers only in the range [TFR1, - TMAX]. NOTE: for small sampling rates TMAX can get smaller than TFR1. - */ - int T0_min, T0_max; - - if (acb_idx < (PIT_FR2 - PIT_MIN) * 4) { - /* first interval with 0.25 pitch resolution */ - T0 = PIT_MIN + (acb_idx / 4); - T0_frac = acb_idx & 0x3; - } else if (acb_idx < ((PIT_FR2 - PIT_MIN) * 4 + (PIT_FR1 - PIT_FR2) * 2)) { - /* second interval with 0.5 pitch resolution */ - acb_idx -= (PIT_FR2 - PIT_MIN) * 4; - T0 = PIT_FR2 + (acb_idx / 2); - T0_frac = (acb_idx & 0x1) * 2; - } else { - /* third interval with 1.0 pitch resolution */ - T0 = acb_idx + PIT_FR1 - ((PIT_FR2 - PIT_MIN) * 4) - - ((PIT_FR1 - PIT_FR2) * 2); - T0_frac = 0; - } - /* find T0_min and T0_max for subframe 1 or 3 */ - T0_min = T0 - 8; - if (T0_min < PIT_MIN) { - T0_min = PIT_MIN; - } - T0_max = T0_min + 15; - if (T0_max > PIT_MAX) { - T0_max = PIT_MAX; - T0_min = T0_max - 15; - } - *pT0_min = T0_min; - *pT0_max = T0_max; - } - *pT0 = T0; - *pT0_frac = T0_frac; - - return error; -} -static void ConcealPitchLag(CAcelpStaticMem *acelp_mem, const int PIT_MAX, - int *pT0, int *pT0_frac) { - USHORT *pold_T0 = &acelp_mem->old_T0; - UCHAR *pold_T0_frac = &acelp_mem->old_T0_frac; - - if ((int)*pold_T0 >= PIT_MAX) { - *pold_T0 = (UCHAR)(PIT_MAX - 5); - } - *pT0 = (int)*pold_T0; - *pT0_frac = (int)*pold_T0_frac; -} - -static UCHAR tab_coremode2nbits[8] = {20, 28, 36, 44, 52, 64, 12, 16}; - -static int MapCoreMode2NBits(int core_mode) { - return (int)tab_coremode2nbits[core_mode]; -} - -void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset, - const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], - const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], - FIXP_SGL stab_fac, CAcelpChannelData *pAcelpData, - INT numLostSubframes, int lastLpcLost, int frameCnt, - FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain, - INT coreCoderFrameLength) { - int i_subfr, subfr_nr, l_div, T; - int T0 = -1, T0_frac = -1; /* mark invalid */ - - int pit_gain_index = 0; - - const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); /* maximum pitch lag */ - - FIXP_COD *code; - FIXP_DBL *exc2; - FIXP_DBL *syn; - FIXP_DBL *exc; - FIXP_LPC A[M_LP_FILTER_ORDER]; - INT A_exp; - - FIXP_DBL period_fac; - FIXP_SGL gain_pit; - FIXP_DBL gain_code, gain_code_smooth, Ener_code; - int Ener_code_e; - int n; - int bfi = (numLostSubframes > 0) ? 1 : 0; - - C_ALLOC_SCRATCH_START( - exc_buf, FIXP_DBL, - PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); /* 411 + 17 + 256 + 1 = 685 */ - C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, - M_LP_FILTER_ORDER + L_DIV); /* 16 + 256 = 272 */ - /* use same memory for code[L_SUBFR] and exc2[L_SUBFR] */ - C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_SUBFR); /* 64 */ - /* make sure they don't overlap if they are accessed alternatingly in - * BuildAdaptiveExcitation() */ -#if (COD_BITS == FRACT_BITS) - code = (FIXP_COD *)(tmp_buf + L_SUBFR / 2); -#elif (COD_BITS == DFRACT_BITS) - code = (FIXP_COD *)tmp_buf; -#endif - exc2 = (FIXP_DBL *)tmp_buf; - - syn = syn_buf + M_LP_FILTER_ORDER; - exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; - - FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, - M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); - FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, - (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); - - FDKmemclear(exc_buf + (PIT_MAX_MAX + L_INTERPOL), - (L_DIV + 1) * sizeof(FIXP_DBL)); - - l_div = coreCoderFrameLength / NB_DIV; - - for (i_subfr = 0, subfr_nr = 0; i_subfr < l_div; - i_subfr += L_SUBFR, subfr_nr++) { - /*-------------------------------------------------* - * - Decode pitch lag (T0 and T0_frac) * - *-------------------------------------------------*/ - if (bfi) { - ConcealPitchLag(acelp_mem, PIT_MAX, &T0, &T0_frac); - } else { - T0 = (int)pAcelpData->T0[subfr_nr]; - T0_frac = (int)pAcelpData->T0_frac[subfr_nr]; - } - - /*-------------------------------------------------* - * - Find the pitch gain, the interpolation filter * - * and the adaptive codebook vector. * - *-------------------------------------------------*/ - Pred_lt4(&exc[i_subfr], T0, T0_frac); - - if ((!bfi && pAcelpData->ltp_filtering_flag[subfr_nr] == 0) || - (bfi && numLostSubframes == 1 && stab_fac < FL2FXCONST_SGL(0.25f))) { - /* find pitch excitation with lp filter: v'(n) => v(n) */ - Pred_lt4_postfilter(&exc[i_subfr]); - } - - /*-------------------------------------------------------* - * - Decode innovative codebook. * - * - Add the fixed-gain pitch contribution to code[]. * - *-------------------------------------------------------*/ - if (bfi) { - for (n = 0; n < L_SUBFR; n++) { - code[n] = - FX_SGL2FX_COD((FIXP_SGL)E_UTIL_random(&acelp_mem->seed_ace)) >> 4; - } - } else { - int nbits = MapCoreMode2NBits((int)pAcelpData->acelp_core_mode); - D_ACELP_decode_4t64(pAcelpData->icb_index[subfr_nr], nbits, &code[0]); - } - - T = T0; - if (T0_frac > 2) { - T += 1; - } - - Preemph_code(code); - Pit_shrp(code, T); - - /* Output pitch lag for bass post-filter */ - if (T > PIT_MAX) { - pT[subfr_nr] = PIT_MAX; - } else { - pT[subfr_nr] = T; - } - D_gain2_plus( - pAcelpData->gains[subfr_nr], - code, /* (i) : Innovative code vector, exponent = SF_CODE */ - &gain_pit, /* (o) : Quantized pitch gain, exponent = SF_GAIN_P */ - &gain_code, /* (o) : Quantized codebook gain */ - pAcelpData - ->mean_energy, /* (i) : mean_ener defined in open-loop (2 bits) */ - bfi, &acelp_mem->past_gpit, &acelp_mem->past_gcode, - &Ener_code, /* (o) : Innovative code vector energy */ - &Ener_code_e); /* (o) : Innovative code vector energy exponent */ - - pit_gain[pit_gain_index++] = FX_SGL2FX_DBL(gain_pit); - - /* calc periodicity factor r_v */ - period_fac = - calc_period_factor(/* (o) : factor (-1=unvoiced to 1=voiced) */ - &exc[i_subfr], /* (i) : pitch excitation, exponent = - SF_EXC */ - gain_pit, /* (i) : gain of pitch, exponent = - SF_GAIN_P */ - gain_code, /* (i) : gain of code */ - Ener_code, /* (i) : Energy of code[] */ - Ener_code_e); /* (i) : Exponent of energy of code[] - */ - - if (lastLpcLost && frameCnt == 0) { - if (gain_pit > FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P))) { - gain_pit = FL2FXCONST_SGL(1.0f / (1 << SF_GAIN_P)); - } - } - - gain_code_smooth = - noise_enhancer(/* (o) : smoothed gain g_sc exponent = SF_GAIN_C */ - gain_code, /* (i) : Quantized codebook gain */ - period_fac, /* (i) : periodicity factor (-1=unvoiced to - 1=voiced) */ - stab_fac, /* (i) : stability factor (0 <= ... < 1), - exponent = 1 */ - &acelp_mem->gc_threshold); - - /* Compute adaptive codebook update u'(n), pitch enhancement c'(n) and - * post-processed excitation u(n). */ - BuildAdaptiveExcitation(code, exc + i_subfr, gain_pit, gain_code, - gain_code_smooth, period_fac, exc2); - - /* Interpolate filter coeffs for current subframe in lsp domain and convert - * to LP domain */ - int_lpc_acelp(lsp_old, /* input : LSPs from past frame */ - lsp_new, /* input : LSPs from present frame */ - subfr_nr, /* input : ACELP subframe index */ - coreCoderFrameLength / L_DIV, - A, /* output: LP coefficients of this subframe */ - &A_exp); - - Syn_filt(A, /* (i) : a[m] prediction coefficients */ - A_exp, L_SUBFR, /* (i) : length */ - exc2, /* (i) : input signal */ - &syn[i_subfr] /* (i/o) : filter states / output signal */ - ); - - } /* end of subframe loop */ - - /* update pitch value for bfi procedure */ - acelp_mem->old_T0_frac = T0_frac; - acelp_mem->old_T0 = T0; - - /* save old excitation and old synthesis memory for next ACELP frame */ - FDKmemcpy(acelp_mem->old_exc_mem, exc + l_div - (PIT_MAX_MAX + L_INTERPOL), - sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); - FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + l_div, - sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); - - Deemph(syn, synth, l_div, - &acelp_mem->de_emph_mem); /* ref soft: mem = synth[-1] */ - - scaleValues(synth, l_div, -ACELP_OUTSCALE); - acelp_mem->deemph_mem_wsyn = acelp_mem->de_emph_mem; - - C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_SUBFR); - C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); - C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV + 1); - return; -} - -void CLpd_AcelpReset(CAcelpStaticMem *acelp) { - acelp->gc_threshold = (FIXP_DBL)0; - - acelp->past_gpit = (FIXP_SGL)0; - acelp->past_gcode = (FIXP_DBL)0; - acelp->old_T0 = 64; - acelp->old_T0_frac = 0; - acelp->deemph_mem_wsyn = (FIXP_DBL)0; - acelp->wsyn_rms = (FIXP_DBL)0; - acelp->seed_ace = 0; -} - -/* TCX time domain concealment */ -/* Compare to figure 13a on page 54 in 3GPP TS 26.290 */ -void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch, - const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], - const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], - const FIXP_SGL stab_fac, INT nLostSf, FIXP_DBL synth[], - INT coreCoderFrameLength, UCHAR last_tcx_noise_factor) { - /* repeat past excitation with pitch from previous decoded TCX frame */ - C_ALLOC_SCRATCH_START( - exc_buf, FIXP_DBL, - PIT_MAX_MAX + L_INTERPOL + L_DIV); /* 411 + 17 + 256 + 1 = */ - C_ALLOC_SCRATCH_START(syn_buf, FIXP_DBL, - M_LP_FILTER_ORDER + L_DIV); /* 256 + 16 = */ - /* += */ - FIXP_DBL ns_buf[L_DIV + 1]; - FIXP_DBL *syn = syn_buf + M_LP_FILTER_ORDER; - FIXP_DBL *exc = exc_buf + PIT_MAX_MAX + L_INTERPOL; - FIXP_DBL *ns = ns_buf + 1; - FIXP_DBL tmp, fact_exc; - INT T = fMin(*pitch, (SHORT)PIT_MAX_MAX); - int i, i_subfr, subfr_nr; - int lDiv = coreCoderFrameLength / NB_DIV; - - FDKmemcpy(syn_buf, acelp_mem->old_syn_mem, - M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); - FDKmemcpy(exc_buf, acelp_mem->old_exc_mem, - (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); - - /* if we lost all packets (i.e. 1 packet of TCX-20 ms, 2 packets of - the TCX-40 ms or 4 packets of the TCX-80ms), we lost the whole - coded frame extrapolation strategy: repeat lost excitation and - use extrapolated LSFs */ - - /* AMR-WB+ like TCX TD concealment */ - - /* number of lost frame cmpt */ - if (nLostSf < 2) { - fact_exc = FL2FXCONST_DBL(0.8f); - } else { - fact_exc = FL2FXCONST_DBL(0.4f); - } - - /* repeat past excitation */ - for (i = 0; i < lDiv; i++) { - exc[i] = fMult(fact_exc, exc[i - T]); - } - - tmp = fMult(fact_exc, acelp_mem->wsyn_rms); - acelp_mem->wsyn_rms = tmp; - - /* init deemph_mem_wsyn */ - acelp_mem->deemph_mem_wsyn = exc[-1]; - - ns[-1] = acelp_mem->deemph_mem_wsyn; - - for (i_subfr = 0, subfr_nr = 0; i_subfr < lDiv; - i_subfr += L_SUBFR, subfr_nr++) { - FIXP_DBL tRes[L_SUBFR]; - FIXP_LPC A[M_LP_FILTER_ORDER]; - INT A_exp; - - /* interpolate LPC coefficients */ - int_lpc_acelp(lsp_old, lsp_new, subfr_nr, lDiv / L_SUBFR, A, &A_exp); - - Syn_filt(A, /* (i) : a[m] prediction coefficients */ - A_exp, L_SUBFR, /* (i) : length */ - &exc[i_subfr], /* (i) : input signal */ - &syn[i_subfr] /* (i/o) : filter states / output signal */ - ); - - E_LPC_a_weight( - A, A, - M_LP_FILTER_ORDER); /* overwrite A as it is not needed any longer */ - - E_UTIL_residu(A, A_exp, &syn[i_subfr], tRes, L_SUBFR); - - Deemph(tRes, &ns[i_subfr], L_SUBFR, &acelp_mem->deemph_mem_wsyn); - - /* Amplitude limiter (saturate at wsyn_rms) */ - for (i = i_subfr; i < i_subfr + L_SUBFR; i++) { - if (ns[i] > tmp) { - ns[i] = tmp; - } else { - if (ns[i] < -tmp) { - ns[i] = -tmp; - } - } - } - - E_UTIL_preemph(&ns[i_subfr], tRes, L_SUBFR); - - Syn_filt(A, /* (i) : a[m] prediction coefficients */ - A_exp, L_SUBFR, /* (i) : length */ - tRes, /* (i) : input signal */ - &syn[i_subfr] /* (i/o) : filter states / output signal */ - ); - - FDKmemmove(&synth[i_subfr], &syn[i_subfr], L_SUBFR * sizeof(FIXP_DBL)); - } - - /* save old excitation and old synthesis memory for next ACELP frame */ - FDKmemcpy(acelp_mem->old_exc_mem, exc + lDiv - (PIT_MAX_MAX + L_INTERPOL), - sizeof(FIXP_DBL) * (PIT_MAX_MAX + L_INTERPOL)); - FDKmemcpy(acelp_mem->old_syn_mem, syn_buf + lDiv, - sizeof(FIXP_DBL) * M_LP_FILTER_ORDER); - acelp_mem->de_emph_mem = acelp_mem->deemph_mem_wsyn; - - C_ALLOC_SCRATCH_END(syn_buf, FIXP_DBL, M_LP_FILTER_ORDER + L_DIV); - C_ALLOC_SCRATCH_END(exc_buf, FIXP_DBL, PIT_MAX_MAX + L_INTERPOL + L_DIV); -} - -void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, - INT *old_T_pf, FIXP_DBL *pit_gain, - FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset, - INT coreCoderFrameLength, INT synSfd, - INT nbSubfrSuperfr) { - int n; - - /* init beginning of synth_buf with old synthesis from previous frame */ - FDKmemcpy(synth_buf, old_synth, sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); - - /* calculate pitch lag offset for ACELP decoder */ - *i_offset = - (samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - - PIT_MIN_12k8; - - /* for bass postfilter */ - for (n = 0; n < synSfd; n++) { - pitch[n] = old_T_pf[n]; - pit_gain[n] = old_gain_pf[n]; - } - for (n = 0; n < nbSubfrSuperfr; n++) { - pitch[n + synSfd] = L_SUBFR; - pit_gain[n + synSfd] = (FIXP_DBL)0; - } -} - -void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, - INT *old_T_pf, INT coreCoderFrameLength, INT synSfd, - INT nbSubfrSuperfr) { - int n; - - /* store last part of synth_buf (which is not handled by the IMDCT overlap) - * for next frame */ - FDKmemcpy(old_synth, synth_buf + coreCoderFrameLength, - sizeof(FIXP_DBL) * (PIT_MAX_MAX - BPF_DELAY)); - - /* for bass postfilter */ - for (n = 0; n < synSfd; n++) { - old_T_pf[n] = pitch[nbSubfrSuperfr + n]; - } -} - -#define L_FAC_ZIR (LFAC) - -void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp, - CAcelpStaticMem *acelp_mem, const INT length, - FIXP_DBL zir[], int doDeemph) { - C_ALLOC_SCRATCH_START(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); - FDK_ASSERT(length <= L_FAC_ZIR); - - FDKmemcpy(tmp_buf, acelp_mem->old_syn_mem, - M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); - FDKmemset(tmp_buf + M_LP_FILTER_ORDER, 0, L_FAC_ZIR * sizeof(FIXP_DBL)); - - Syn_filt(A, A_exp, length, &tmp_buf[M_LP_FILTER_ORDER], - &tmp_buf[M_LP_FILTER_ORDER]); - if (!doDeemph) { - /* if last lpd mode was TD concealment, then bypass deemph */ - FDKmemcpy(zir, tmp_buf, length * sizeof(*zir)); - } else { - Deemph(&tmp_buf[M_LP_FILTER_ORDER], &zir[0], length, - &acelp_mem->de_emph_mem); - scaleValues(zir, length, -ACELP_OUTSCALE); - } - C_ALLOC_SCRATCH_END(tmp_buf, FIXP_DBL, L_FAC_ZIR + M_LP_FILTER_ORDER); -} - -void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode, - UCHAR last_last_lpd_mode, - const FIXP_LPC *A_new, const INT A_new_exp, - const FIXP_LPC *A_old, const INT A_old_exp, - CAcelpStaticMem *acelp_mem, - INT coreCoderFrameLength, INT clearOldExc, - UCHAR lpd_mode) { - int l_div = - coreCoderFrameLength / NB_DIV; /* length of one ACELP/TCX20 frame */ - int l_div_partial; - FIXP_DBL *syn, *old_exc_mem; - - C_ALLOC_SCRATCH_START(synth_buf, FIXP_DBL, - PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); - syn = &synth_buf[M_LP_FILTER_ORDER]; - - l_div_partial = PIT_MAX_MAX + L_INTERPOL - l_div; - old_exc_mem = acelp_mem->old_exc_mem; - - if (lpd_mode == 4) { - /* Bypass Domain conversion. TCXTD Concealment does no deemphasis in the - * end. */ - FDKmemcpy( - synth_buf, &synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], - (PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER) * sizeof(FIXP_DBL)); - /* Set deemphasis memory state for TD concealment */ - acelp_mem->deemph_mem_wsyn = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); - } else { - /* convert past [PIT_MAX_MAX+L_INTERPOL+M_LP_FILTER_ORDER] synthesis to - * preemph domain */ - E_UTIL_preemph(&synth[-(PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER)], - synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); - scaleValuesSaturate(synth_buf, PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER, - ACELP_OUTSCALE); - } - - /* Set deemphasis memory state */ - acelp_mem->de_emph_mem = scaleValueSaturate(synth[-1], ACELP_OUTSCALE); - - /* update acelp synth filter memory */ - FDKmemcpy(acelp_mem->old_syn_mem, - &syn[PIT_MAX_MAX + L_INTERPOL - M_LP_FILTER_ORDER], - M_LP_FILTER_ORDER * sizeof(FIXP_DBL)); - - if (clearOldExc) { - FDKmemclear(old_exc_mem, (PIT_MAX_MAX + L_INTERPOL) * sizeof(FIXP_DBL)); - C_ALLOC_SCRATCH_END(synth_buf, FIXP_DBL, - PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); - return; - } - - /* update past [PIT_MAX_MAX+L_INTERPOL] samples of exc memory */ - if (last_lpd_mode == 1) { /* last frame was TCX20 */ - if (last_last_lpd_mode == 0) { /* ACELP -> TCX20 -> ACELP transition */ - /* Delay valid part of excitation buffer (from previous ACELP frame) by - * l_div samples */ - FDKmemmove(old_exc_mem, old_exc_mem + l_div, - sizeof(FIXP_DBL) * l_div_partial); - } else if (last_last_lpd_mode > 0) { /* TCX -> TCX20 -> ACELP transition */ - E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, l_div_partial); - } - E_UTIL_residu(A_new, A_new_exp, syn + l_div_partial, - old_exc_mem + l_div_partial, l_div); - } else { /* prev frame was FD, TCX40 or TCX80 */ - int exc_A_new_length = (coreCoderFrameLength / 2 > PIT_MAX_MAX + L_INTERPOL) - ? PIT_MAX_MAX + L_INTERPOL - : coreCoderFrameLength / 2; - int exc_A_old_length = PIT_MAX_MAX + L_INTERPOL - exc_A_new_length; - E_UTIL_residu(A_old, A_old_exp, syn, old_exc_mem, exc_A_old_length); - E_UTIL_residu(A_new, A_new_exp, &syn[exc_A_old_length], - &old_exc_mem[exc_A_old_length], exc_A_new_length); - } - C_ALLOC_SCRATCH_END(synth_buf, FIXP_DBL, - PIT_MAX_MAX + L_INTERPOL + M_LP_FILTER_ORDER); - - return; -} - -FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length) { - FDK_ASSERT(length <= PIT_MAX_MAX + L_INTERPOL); - return acelp_mem->old_exc_mem; -} - -INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelp, - INT acelp_core_mode, INT coreCoderFrameLength, - INT i_offset) { - int nb_subfr = coreCoderFrameLength / L_DIV; - const UCHAR *num_acb_index_bits = - (nb_subfr == 4) ? num_acb_idx_bits_table[0] : num_acb_idx_bits_table[1]; - int nbits; - int error = 0; - - const int PIT_MIN = PIT_MIN_12k8 + i_offset; - const int PIT_FR2 = PIT_FR2_12k8 - i_offset; - const int PIT_FR1 = PIT_FR1_12k8; - const int PIT_MAX = PIT_MAX_12k8 + (6 * i_offset); - int T0, T0_frac, T0_min = 0, T0_max; - - if (PIT_MAX > PIT_MAX_MAX) { - error = AAC_DEC_DECODE_FRAME_ERROR; - goto bail; - } - - acelp->acelp_core_mode = acelp_core_mode; - - nbits = MapCoreMode2NBits(acelp_core_mode); - - /* decode mean energy with 2 bits : 18, 30, 42 or 54 dB */ - acelp->mean_energy = FDKreadBits(hBs, 2); - - for (int sfr = 0; sfr < nb_subfr; sfr++) { - /* read ACB index and store T0 and T0_frac for each ACELP subframe. */ - error = DecodePitchLag(hBs, num_acb_index_bits[sfr], PIT_MIN, PIT_FR2, - PIT_FR1, PIT_MAX, &T0, &T0_frac, &T0_min, &T0_max); - if (error) { - goto bail; - } - acelp->T0[sfr] = (USHORT)T0; - acelp->T0_frac[sfr] = (UCHAR)T0_frac; - acelp->ltp_filtering_flag[sfr] = FDKreadBits(hBs, 1); - switch (nbits) { - case 12: /* 12 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 1); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); - break; - case 16: /* 16 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 1); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); - break; - case 20: /* 20 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); - break; - case 28: /* 28 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 5); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 5); - break; - case 36: /* 36 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); - break; - case 44: /* 44 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 9); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 9); - break; - case 52: /* 52 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 13); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 13); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 13); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 13); - break; - case 64: /* 64 bits AMR-WB codebook is used */ - acelp->icb_index[sfr][0] = FDKreadBits(hBs, 2); - acelp->icb_index[sfr][1] = FDKreadBits(hBs, 2); - acelp->icb_index[sfr][2] = FDKreadBits(hBs, 2); - acelp->icb_index[sfr][3] = FDKreadBits(hBs, 2); - acelp->icb_index[sfr][4] = FDKreadBits(hBs, 14); - acelp->icb_index[sfr][5] = FDKreadBits(hBs, 14); - acelp->icb_index[sfr][6] = FDKreadBits(hBs, 14); - acelp->icb_index[sfr][7] = FDKreadBits(hBs, 14); - break; - default: - FDK_ASSERT(0); - break; - } - acelp->gains[sfr] = FDKreadBits(hBs, 7); - } - -bail: - return error; -} diff --git a/libAACdec/src/usacdec_acelp.h b/libAACdec/src/usacdec_acelp.h deleted file mode 100644 index 9de41ff..0000000 --- a/libAACdec/src/usacdec_acelp.h +++ /dev/null @@ -1,281 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand - - Description: USAC ACELP frame decoder - -*******************************************************************************/ - -#ifndef USACDEC_ACELP_H -#define USACDEC_ACELP_H - -#include "common_fix.h" -#include "FDK_bitstream.h" -#include "usacdec_const.h" -#include "usacdec_rom.h" - -//#define ENHANCED_TCX_TD_CONCEAL_ENABLE - -/** Structure which holds the ACELP internal persistent memory */ -typedef struct { - FIXP_DBL old_exc_mem[PIT_MAX_MAX + L_INTERPOL]; - FIXP_DBL old_syn_mem[M_LP_FILTER_ORDER]; /* synthesis filter states */ - FIXP_SGL A[M_LP_FILTER_ORDER]; - INT A_exp; - FIXP_DBL gc_threshold; - FIXP_DBL de_emph_mem; - FIXP_SGL past_gpit; - FIXP_DBL past_gcode; - USHORT old_T0; - UCHAR old_T0_frac; - FIXP_DBL deemph_mem_wsyn; - FIXP_DBL wsyn_rms; - SHORT seed_ace; -} CAcelpStaticMem; - -/** Structure which holds the parameter data needed to decode one ACELP frame. - */ -typedef struct { - UCHAR - acelp_core_mode; /**< mean excitation energy index for whole ACELP frame - */ - UCHAR mean_energy; /**< acelp core mode for whole ACELP frame */ - USHORT T0[NB_SUBFR]; - UCHAR T0_frac[NB_SUBFR]; - UCHAR ltp_filtering_flag[NB_SUBFR]; /**< controlls whether LTP postfilter is - active for each ACELP subframe */ - SHORT icb_index[NB_SUBFR] - [8]; /**< innovative codebook index for each ACELP subframe */ - UCHAR gains[NB_SUBFR]; /**< gain index for each ACELP subframe */ -} CAcelpChannelData; - -/** - * \brief Read the acelp_coding() bitstream part. - * \param[in] hBs bitstream handle to read data from. - * \param[out] acelpData pointer to structure to store the parsed data of one - * ACELP frame. - * \param[in] acelp_core_mode the ACELP core mode index. - * \param[in] coreCoderFrameLength length of core coder frame (1024|768) - */ -INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelpData, - INT acelp_core_mode, INT i_offset, INT coreCoderFrameLength); -/** - * \brief Initialization of memory before one LPD frame is decoded - * \param[out] synth_buf synthesis buffer to be initialized, exponent = SF_SYNTH - * \param[in] old_synth past synthesis of previous LPD frame, exponent = - * SF_SYNTH - * \param[out] synth_buf_fb fullband synthesis buffer to be initialized, - * exponent = SF_SYNTH - * \param[in] old_synth_fb past fullband synthesis of previous LPD frame, - * exponent = SF_SYNTH - * \param[out] pitch vector where decoded pitch lag values are stored - * \param[in] old_T_pf past pitch lag values of previous LPD frame - * \param[in] samplingRate sampling rate for pitch lag offset calculation - * \param[out] i_offset pitch lag offset for the decoding of the pitch lag - * \param[in] coreCoderFrameLength length of core coder frame (1024|768) - */ -void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, - INT *old_T_pf, FIXP_DBL *pit_gain, - FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset, - INT coreCoderFrameLength, INT synSfd, - INT nbSubfrSuperfr); - -/** - * \brief Save tail of buffers for the initialization of the next LPD frame - * \param[in] synth_buf synthesis of current LPD frame, exponent = SF_SYNTH - * \param[out] old_synth memory where tail of fullband synth_buf is stored, - * exponent = SF_SYNTH - * \param[in] synth_buf_fb fullband synthesis of current LPD frame, exponent = - * SF_SYNTH - * \param[out] old_synth_fb memory where tail of fullband synth_buf is stored, - * exponent = SF_SYNTH - * \param[in] pitch decoded pitch lag values of current LPD frame - * \param[out] old_T_pf memory where last SYN_SFD pitch lag values are stored - */ -void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch, - INT *old_T_pf, INT coreCoderFrameLength, INT synSfd, - INT nbSubfrSuperfr); - -/** - * \brief Decode one ACELP frame (three or four ACELP subframes with 64 samples - * each) - * \param[in,out] acelp_mem pointer to ACELP memory structure - * \param[in] i_offset pitch lag offset - * \param[in] lsp_old LPC filter in LSP domain corresponding to previous frame - * \param[in] lsp_new LPC filter in LSP domain corresponding to current frame - * \param[in] stab_fac stability factor constrained by 0<=stab_fac<=1.0, - * exponent = SF_STAB - * \param[in] acelpData pointer to struct with data which is needed for decoding - * one ACELP frame - * \param[out] synth ACELP output signal - * \param[out] pT four decoded pitch lag values - * \param[in] coreCoderFrameLength length of core coder frame (1024|768) - */ -void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset, - const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], - const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], - FIXP_SGL stab_fac, CAcelpChannelData *acelpData, - INT numLostSubframes, int lastLpcLost, int frameCnt, - FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain, - INT coreCoderFrameLength); - -/** - * \brief Reset ACELP internal memory. - * \param[out] acelp_mem pointer to ACELP memory structure - */ -void CLpd_AcelpReset(CAcelpStaticMem *acelp_mem); - -/** - * \brief Initialize ACELP internal memory in case of FAC before ACELP decoder - * is called - * \param[in] synth points to end+1 of past valid synthesis signal, exponent = - * SF_SYNTH - * \param[in] last_lpd_mode last lpd mode - * \param[in] last_last_lpd_mode lpd mode before last_lpd_mode - * \param[in] A_new LP synthesis filter coeffs corresponding to last frame, - * exponent = SF_A_COEFFS - * \param[in] A_old LP synthesis filter coeffs corresponding to the frame before - * last frame, exponent = SF_A_COEFFS - * \param[in,out] acelp_mem pointer to ACELP memory structure - * \param[in] coreCoderFrameLength length of core coder frame (1024|768) - */ -void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode, - UCHAR last_last_lpd_mode, - const FIXP_LPC *A_new, const INT A_new_exp, - const FIXP_LPC *A_old, const INT A_old_exp, - CAcelpStaticMem *acelp_mem, - INT coreCoderFrameLength, INT clearOldExc, - UCHAR lpd_mode); - -/** - * \brief Calculate zero input response (zir) of the acelp synthesis filter - * \param[in] A LP synthesis filter coefficients, exponent = SF_A_COEFFS - * \param[in,out] acelp_mem pointer to ACELP memory structure - * \param[in] length length of zir - * \param[out] zir pointer to zir output buffer, exponent = SF_SYNTH - */ -void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp, - CAcelpStaticMem *acelp_mem, const INT length, - FIXP_DBL zir[], int doDeemph); - -/** - * \brief Borrow static excitation memory from ACELP decoder - * \param[in] acelp_mem pointer to ACELP memory structure - * \param[in] length number of requested FIXP_DBL values - * \return pointer to requested memory - * - * The caller has to take care not to overwrite valid memory areas. - * During TCX/FAC calculations and before CLpd_AcelpPrepareInternalMem() is - * called, the following memory size is available: - * - 256 samples in case of ACELP -> TCX20 -> ACELP transition - * - PIT_MAX_MAX+L_INTERPOL samples in all other cases - */ -FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length); - -void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch, - const FIXP_LPC lsp_old[M_LP_FILTER_ORDER], - const FIXP_LPC lsp_new[M_LP_FILTER_ORDER], - const FIXP_SGL stab_fac, INT numLostSubframes, - FIXP_DBL synth[], INT coreCoderFrameLength, - UCHAR last_tcx_noise_factor); - -inline SHORT E_UTIL_random(SHORT *seed) { - *seed = (SHORT)((((LONG)*seed * (LONG)31821) >> 1) + (LONG)13849); - return (*seed); -} - -#endif /* USACDEC_ACELP_H */ diff --git a/libAACdec/src/usacdec_const.h b/libAACdec/src/usacdec_const.h deleted file mode 100644 index f68e808..0000000 --- a/libAACdec/src/usacdec_const.h +++ /dev/null @@ -1,203 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Manuel Jander - - Description: USAC related constants - -*******************************************************************************/ - -#ifndef USACDEC_CONST_H -#define USACDEC_CONST_H - -/* scale factors */ -#define SF_CODE 6 /* exponent of code[], fixed codebook vector */ -#define SF_GAIN_C 16 /* exponent of gain code and smoothed gain code */ -#define SF_EXC 16 /* exponent of exc[] and exc2[], excitation buffer */ -#define SF_GAIN_P 1 /* exponent of gain_pit */ -#define SF_PFAC 0 /* exponent of period/voicing factor */ -#define SF_SYNTH SF_EXC /* exponent of synthesis buffer */ -#define SF_A_COEFFS 3 /* exponent of LP domain synthesis filter coefficient */ -#define SF_STAB 1 /* exponent of stability factor */ - -/* definitions which are independent of coreCoderFrameLength */ -#define M_LP_FILTER_ORDER 16 /* LP filter order */ -#define LP_FILTER_SCALE 4 /* LP filter scale */ - -#define PIT_MIN_12k8 34 /* Minimum pitch lag with resolution 1/4 */ -#define PIT_MAX_12k8 231 /* Maximum pitch lag for fs=12.8kHz */ -#define FSCALE_DENOM 12800 /* Frequency scale denominator */ -#define FAC_FSCALE_MIN \ - 6000 /* Minimum allowed frequency scale for acelp decoder */ - -#if !defined(LPD_MAX_CORE_SR) -#define LPD_MAX_CORE_SR 24000 /* Default value from ref soft */ -#endif -#define FAC_FSCALE_MAX \ - LPD_MAX_CORE_SR /* Maximum allowed frequency scale for acelp decoder */ - -/* Maximum pitch lag (= 411 for fs_max = 24000) */ -#define PIT_MAX_TMP \ - (PIT_MAX_12k8 + \ - (6 * \ - ((((FAC_FSCALE_MAX * PIT_MIN_12k8) + (FSCALE_DENOM / 2)) / FSCALE_DENOM) - \ - PIT_MIN_12k8))) -#if (PIT_MAX_TMP < \ - 256) /* cannot be smaller because of tcx time domain concealment */ -#define PIT_MAX_MAX 256 -#else -#define PIT_MAX_MAX PIT_MAX_TMP -#endif - -#define NB_DIV 4 /* number of division (20ms) per 80ms frame */ -#define L_SUBFR 64 /* subframe size (5ms) */ -#define BPF_SFD 1 /* bass postfilter delay (subframe) */ -#define BPF_DELAY (BPF_SFD * L_SUBFR) /* bass postfilter delay (samples) */ - -#define L_FILT 12 /* Delay of up-sampling filter (bass post-filter) */ -#define L_EXTRA 96 /* for bass post-filter */ -#define L_INTERPOL \ - (16 + 1) /* Length of filter for interpolation (acelp decoder) */ - -/* definitions for coreCoderFrameLength = 1024 */ -#define L_FRAME_PLUS_1024 1024 /* length of one 80ms superframe */ -#define L_DIV_1024 \ - (L_FRAME_PLUS_1024 / NB_DIV) /* length of one acelp or tcx20 frame */ -#define NB_SUBFR_1024 \ - (L_DIV_1024 / L_SUBFR) /* number of 5ms subframe per division */ -#define NB_SUBFR_SUPERFR_1024 \ - (L_FRAME_PLUS_1024 / L_SUBFR) /* number of 5ms subframe per 80ms frame */ -#define AAC_SFD_1024 (NB_SUBFR_SUPERFR_1024 / 2) /* AAC delay (subframe) */ -#define AAC_DELAY_1024 (AAC_SFD_1024 * L_SUBFR) /* AAC delay (samples) */ -#define SYN_SFD_1024 (AAC_SFD_1024 - BPF_SFD) /* synthesis delay (subframe) */ -#define SYN_DELAY_1024 \ - (SYN_SFD_1024 * L_SUBFR) /* synthesis delay (samples) \ - */ -#define LFAC_1024 (L_DIV_1024 / 2) /* FAC frame length */ -#define LFAC_SHORT_1024 \ - (L_DIV_1024 / 4) /* for transitions EIGHT_SHORT FD->LPD and vv. */ -#define FDNS_NPTS_1024 64 /* FD noise shaping resolution (64=100Hz/point) */ - -/* definitions for coreCoderFrameLength = 768 */ -#define L_FRAME_PLUS_768 768 -#define L_DIV_768 \ - (L_FRAME_PLUS_768 / NB_DIV) /* length of one acelp or tcx20 frame */ -#define NB_SUBFR_768 \ - (L_DIV_768 / L_SUBFR) /* number of 5ms subframe per division */ -#define NB_SUBFR_SUPERFR_768 \ - (L_FRAME_PLUS_768 / L_SUBFR) /* number of 5ms subframe per 80ms frame */ -#define AAC_SFD_768 (NB_SUBFR_SUPERFR_768 / 2) /* AAC delay (subframe) */ -#define AAC_DELAY_768 (AAC_SFD_768 * L_SUBFR) /* AAC delay (samples) */ -#define SYN_SFD_768 (AAC_SFD_768 - BPF_SFD) /* synthesis delay (subframe) */ -#define SYN_DELAY_768 (SYN_SFD_768 * L_SUBFR) /* synthesis delay (samples) */ -#define LFAC_768 (L_DIV_768 / 2) /* FAC frame length */ -#define LFAC_SHORT_768 \ - (L_DIV_768 / 4) /* for transitions EIGHT_SHORT FD->LPD and vv. */ - -/* maximum (used for memory allocation) */ -#define L_FRAME_PLUS L_FRAME_PLUS_1024 -#define L_DIV L_DIV_1024 -#define NB_SUBFR NB_SUBFR_1024 -#define NB_SUBFR_SUPERFR NB_SUBFR_SUPERFR_1024 -#define AAC_SFD AAC_SFD_1024 -#define AAC_DELAY AAC_DELAY_1024 -#define SYN_SFD SYN_SFD_1024 -#define SYN_DELAY SYN_DELAY_1024 -#define LFAC LFAC_1024 -#define LFAC_SHORT LFAC_SHORT_1024 -#define FDNS_NPTS FDNS_NPTS_1024 - -#endif /* USACDEC_CONST_H */ diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp deleted file mode 100644 index 0d3d844..0000000 --- a/libAACdec/src/usacdec_fac.cpp +++ /dev/null @@ -1,745 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Manuel Jander - - Description: USAC FAC - -*******************************************************************************/ - -#include "usacdec_fac.h" - -#include "usacdec_const.h" -#include "usacdec_lpc.h" -#include "usacdec_acelp.h" -#include "usacdec_rom.h" -#include "dct.h" -#include "FDK_tools_rom.h" -#include "mdct.h" - -#define SPEC_FAC(ptr, i, gl) ((ptr) + ((i) * (gl))) - -FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo, - UCHAR mod[NB_DIV], int *pState) { - FIXP_DBL *ptr; - int i; - int k = 0; - int max_windows = 8; - - FDK_ASSERT(*pState >= 0 && *pState < max_windows); - - /* Look for free space to store FAC data. 2 FAC data blocks fit into each TCX - * spectral data block. */ - for (i = *pState; i < max_windows; i++) { - if (mod[i >> 1] == 0) { - break; - } - } - - *pState = i + 1; - - if (i == max_windows) { - ptr = pAacDecoderChannelInfo->data.usac.fac_data0; - } else { - FDK_ASSERT(mod[(i >> 1)] == 0); - ptr = SPEC_FAC(pAacDecoderChannelInfo->pSpectralCoefficient, i, - pAacDecoderChannelInfo->granuleLength << k); - } - - return ptr; -} - -int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale, - int length, int use_gain, int frame) { - FIXP_DBL fac_gain; - int fac_gain_e = 0; - - if (use_gain) { - CLpd_DecodeGain(&fac_gain, &fac_gain_e, FDKreadBits(hBs, 7)); - } - - if (CLpc_DecodeAVQ(hBs, pFac, 1, 1, length) != 0) { - return -1; - } - - { - int scale; - - scale = getScalefactor(pFac, length); - scaleValues(pFac, length, scale); - pFacScale[frame] = DFRACT_BITS - 1 - scale; - } - - if (use_gain) { - int i; - - pFacScale[frame] += fac_gain_e; - - for (i = 0; i < length; i++) { - pFac[i] = fMult(pFac[i], fac_gain); - } - } - return 0; -} - -/** - * \brief Apply synthesis filter with zero input to x. The overall filter gain - * is 1.0. - * \param a LPC filter coefficients. - * \param length length of the input/output data vector x. - * \param x input/output vector, where the synthesis filter is applied in place. - */ -static void Syn_filt_zero(const FIXP_LPC a[], const INT a_exp, INT length, - FIXP_DBL x[]) { - int i, j; - FIXP_DBL L_tmp; - - for (i = 0; i < length; i++) { - L_tmp = (FIXP_DBL)0; - - for (j = 0; j < fMin(i, M_LP_FILTER_ORDER); j++) { - L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]) >> (LP_FILTER_SCALE - 1); - } - - L_tmp = scaleValue(L_tmp, a_exp + LP_FILTER_SCALE); - x[i] = fAddSaturate(x[i], L_tmp); - } -} - -/* Table is also correct for coreCoderFrameLength = 768. Factor 3/4 is canceled - out: gainFac = 0.5 * sqrt(fac_length/lFrame) -*/ -static const FIXP_DBL gainFac[4] = {0x40000000, 0x2d413ccd, 0x20000000, - 0x16a09e66}; - -void CFac_ApplyGains(FIXP_DBL fac_data[LFAC], const INT fac_length, - const FIXP_DBL tcx_gain, const FIXP_DBL alfd_gains[], - const INT mod) { - FIXP_DBL facFactor; - int i; - - FDK_ASSERT((fac_length == 128) || (fac_length == 96)); - - /* 2) Apply gain factor to FAC data */ - facFactor = fMult(gainFac[mod], tcx_gain); - for (i = 0; i < fac_length; i++) { - fac_data[i] = fMult(fac_data[i], facFactor); - } - - /* 3) Apply spectrum deshaping using alfd_gains */ - for (i = 0; i < fac_length / 4; i++) { - int k; - - k = i >> (3 - mod); - fac_data[i] = fMult(fac_data[i], alfd_gains[k]) - << 1; /* alfd_gains is scaled by one bit. */ - } -} - -static void CFac_CalcFacSignal(FIXP_DBL *pOut, FIXP_DBL *pFac, - const int fac_scale, const int fac_length, - const FIXP_LPC A[M_LP_FILTER_ORDER], - const INT A_exp, const int fAddZir, - const int isFdFac) { - FIXP_LPC wA[M_LP_FILTER_ORDER]; - FIXP_DBL tf_gain = (FIXP_DBL)0; - int wlength; - int scale = fac_scale; - - /* obtain tranform gain. */ - imdct_gain(&tf_gain, &scale, isFdFac ? 0 : fac_length); - - /* 4) Compute inverse DCT-IV of FAC data. Output scale of DCT IV is 16 bits. - */ - dct_IV(pFac, fac_length, &scale); - /* dct_IV scale = log2(fac_length). "- 7" is a factor of 2/128 */ - if (tf_gain != (FIXP_DBL)0) { /* non-radix 2 transform gain */ - int i; - - for (i = 0; i < fac_length; i++) { - pFac[i] = fMult(tf_gain, pFac[i]); - } - } - scaleValuesSaturate(pOut, pFac, fac_length, - scale); /* Avoid overflow issues and saturate. */ - - E_LPC_a_weight(wA, A, M_LP_FILTER_ORDER); - - /* We need the output of the IIR filter to be longer than "fac_length". - For this reason we run it with zero input appended to the end of the input - sequence, i.e. we generate its ZIR and extend the output signal.*/ - FDKmemclear(pOut + fac_length, fac_length * sizeof(FIXP_DBL)); - wlength = 2 * fac_length; - - /* 5) Apply weighted synthesis filter to FAC data, including optional Zir (5. - * item 4). */ - Syn_filt_zero(wA, A_exp, wlength, pOut); -} - -INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, - const int fac_scale, FIXP_LPC *A, INT A_exp, - INT nrOutSamples, const INT fac_length, - const INT isFdFac, UCHAR prevWindowShape) { - FIXP_DBL *pOvl; - FIXP_DBL *pOut0; - const FIXP_WTP *pWindow; - int i, fl, nrSamples = 0; - - FDK_ASSERT(fac_length <= 1024 / (4 * 2)); - - fl = fac_length * 2; - - pWindow = FDKgetWindowSlope(fl, prevWindowShape); - - /* Adapt window slope length in case of frame loss. */ - if (hMdct->prev_fr != fl) { - int nl = 0; - imdct_adapt_parameters(hMdct, &fl, &nl, fac_length, pWindow, nrOutSamples); - FDK_ASSERT(nl == 0); - } - - if (nrSamples < nrOutSamples) { - pOut0 = output; - nrSamples += hMdct->ov_offset; - /* Purge buffered output. */ - FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); - hMdct->ov_offset = 0; - } - - pOvl = hMdct->overlap.freq + hMdct->ov_size - 1; - - if (nrSamples >= nrOutSamples) { - pOut0 = hMdct->overlap.time + hMdct->ov_offset; - hMdct->ov_offset += hMdct->prev_nr + fl / 2; - } else { - pOut0 = output + nrSamples; - nrSamples += hMdct->prev_nr + fl / 2; - } - if (hMdct->prevPrevAliasSymmetry == 0) { - for (i = 0; i < hMdct->prev_nr; i++) { - FIXP_DBL x = -(*pOvl--); - *pOut0 = IMDCT_SCALE_DBL(x); - pOut0++; - } - } else { - for (i = 0; i < hMdct->prev_nr; i++) { - FIXP_DBL x = (*pOvl--); - *pOut0 = IMDCT_SCALE_DBL(x); - pOut0++; - } - } - hMdct->prev_nr = 0; - - { - if (pFac != NULL) { - /* Note: The FAC gain might have been applied directly after bit stream - * parse in this case. */ - CFac_CalcFacSignal(pOut0, pFac, fac_scale, fac_length, A, A_exp, 0, - isFdFac); - } else { - /* Clear buffer because of the overlap and ADD! */ - FDKmemclear(pOut0, fac_length * sizeof(FIXP_DBL)); - } - } - - i = 0; - - if (hMdct->prevPrevAliasSymmetry == 0) { - for (; i < fl / 2; i++) { - FIXP_DBL x0; - - /* Overlap Add */ - x0 = -fMult(*pOvl--, pWindow[i].v.re); - - *pOut0 += IMDCT_SCALE_DBL(x0); - pOut0++; - } - } else { - for (; i < fl / 2; i++) { - FIXP_DBL x0; - - /* Overlap Add */ - x0 = fMult(*pOvl--, pWindow[i].v.re); - - *pOut0 += IMDCT_SCALE_DBL(x0); - pOut0++; - } - } - if (hMdct->pFacZir != - 0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */ - FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */ - for (i = 0; i < fl / 2; i++) { - pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); - } - hMdct->pFacZir = NULL; - } - - hMdct->prev_fr = 0; - hMdct->prev_nr = 0; - hMdct->prev_tl = 0; - hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; - - return nrSamples; -} - -INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, - const SHORT spec_scale[], const int nSpec, - FIXP_DBL *pFac, const int fac_scale, - const INT fac_length, INT noOutSamples, const INT tl, - const FIXP_WTP *wrs, const INT fr, FIXP_LPC A[16], - INT A_exp, CAcelpStaticMem *acelp_mem, - const FIXP_DBL gain, const int last_frame_lost, - const int isFdFac, const UCHAR last_lpd_mode, - const int k, int currAliasingSymmetry) { - FIXP_DBL *pCurr, *pOvl, *pSpec; - const FIXP_WTP *pWindow; - const FIXP_WTB *FacWindowZir_conceal; - UCHAR doFacZirConceal = 0; - int doDeemph = 1; - const FIXP_WTB *FacWindowZir, *FacWindowSynth; - FIXP_DBL *pOut0 = output, *pOut1; - int w, i, fl, nl, nr, f_len, nrSamples = 0, s = 0, scale, total_gain_e; - FIXP_DBL *pF, *pFAC_and_FAC_ZIR = NULL; - FIXP_DBL total_gain = gain; - - FDK_ASSERT(fac_length <= 1024 / (4 * 2)); - switch (fac_length) { - /* coreCoderFrameLength = 1024 */ - case 128: - pWindow = SineWindow256; - FacWindowZir = FacWindowZir128; - FacWindowSynth = FacWindowSynth128; - break; - case 64: - pWindow = SineWindow128; - FacWindowZir = FacWindowZir64; - FacWindowSynth = FacWindowSynth64; - break; - case 32: - pWindow = SineWindow64; - FacWindowZir = FacWindowZir32; - FacWindowSynth = FacWindowSynth32; - break; - /* coreCoderFrameLength = 768 */ - case 96: - pWindow = SineWindow192; - FacWindowZir = FacWindowZir96; - FacWindowSynth = FacWindowSynth96; - break; - case 48: - pWindow = SineWindow96; - FacWindowZir = FacWindowZir48; - FacWindowSynth = FacWindowSynth48; - break; - default: - FDK_ASSERT(0); - return 0; - } - - FacWindowZir_conceal = FacWindowSynth; - /* Derive NR and NL */ - fl = fac_length * 2; - nl = (tl - fl) >> 1; - nr = (tl - fr) >> 1; - - if (noOutSamples > nrSamples) { - /* Purge buffered output. */ - FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); - nrSamples = hMdct->ov_offset; - hMdct->ov_offset = 0; - } - - if (nrSamples >= noOutSamples) { - pOut1 = hMdct->overlap.time + hMdct->ov_offset; - if (hMdct->ov_offset < fac_length) { - pOut0 = output + nrSamples; - } else { - pOut0 = pOut1; - } - hMdct->ov_offset += fac_length + nl; - } else { - pOut1 = output + nrSamples; - pOut0 = output + nrSamples; - } - - { - pFAC_and_FAC_ZIR = CLpd_ACELP_GetFreeExcMem(acelp_mem, 2 * fac_length); - { - const FIXP_DBL *pTmp1, *pTmp2; - - doFacZirConceal |= ((last_frame_lost != 0) && (k == 0)); - doDeemph &= (last_lpd_mode != 4); - if (doFacZirConceal) { - /* ACELP contribution in concealment case: - Use ZIR with a modified ZIR window to preserve some more energy. - Dont use FAC, which contains wrong information for concealed frame - Dont use last ACELP samples, but double ZIR, instead (afterwards) */ - FDKmemclear(pFAC_and_FAC_ZIR, 2 * fac_length * sizeof(FIXP_DBL)); - FacWindowSynth = (FIXP_WTB *)pFAC_and_FAC_ZIR; - FacWindowZir = FacWindowZir_conceal; - } else { - CFac_CalcFacSignal(pFAC_and_FAC_ZIR, pFac, fac_scale + s, fac_length, A, - A_exp, 1, isFdFac); - } - /* 6) Get windowed past ACELP samples and ACELP ZIR signal */ - - /* - * Get ACELP ZIR (pFac[]) and ACELP past samples (pOut0[]) and add them - * to the FAC synth signal contribution on pOut1[]. - */ - { - { - CLpd_Acelp_Zir(A, A_exp, acelp_mem, fac_length, pFac, doDeemph); - - pTmp1 = pOut0; - pTmp2 = pFac; - } - - for (i = 0, w = 0; i < fac_length; i++) { - FIXP_DBL x; - /* Div2 is compensated by table scaling */ - x = fMultDiv2(pTmp2[i], FacWindowZir[w]); - x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]); - x += pFAC_and_FAC_ZIR[i]; - pOut1[i] = x; - - w++; - } - } - - if (doFacZirConceal) { - /* ZIR is the only ACELP contribution, so double it */ - scaleValues(pOut1, fac_length, 1); - } - } - } - - if (nrSamples < noOutSamples) { - nrSamples += fac_length + nl; - } - - /* Obtain transform gain */ - total_gain = gain; - total_gain_e = 0; - imdct_gain(&total_gain, &total_gain_e, tl); - - /* IMDCT overlap add */ - scale = total_gain_e; - pSpec = _pSpec; - - /* Note:when comming from an LPD frame (TCX/ACELP) the previous alisaing - * symmetry must always be 0 */ - if (currAliasingSymmetry == 0) { - dct_IV(pSpec, tl, &scale); - } else { - FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)]; - FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp); - C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp)); - dst_III(pSpec, tmp, tl, &scale); - C_ALLOC_ALIGNED_UNREGISTER(tmp); - } - - /* Optional scaling of time domain - no yet windowed - of current spectrum */ - if (total_gain != (FIXP_DBL)0) { - for (i = 0; i < tl; i++) { - pSpec[i] = fMult(pSpec[i], total_gain); - } - } - int loc_scale = fixmin_I(spec_scale[0] + scale, (INT)DFRACT_BITS - 1); - scaleValuesSaturate(pSpec, tl, loc_scale); - - pOut1 += fl / 2 - 1; - pCurr = pSpec + tl - fl / 2; - - for (i = 0; i < fl / 2; i++) { - FIXP_DBL x1; - - /* FAC signal is already on pOut1, because of that the += operator. */ - x1 = fMult(*pCurr++, pWindow[i].v.re); - FDK_ASSERT((pOut1 >= hMdct->overlap.time && - pOut1 < hMdct->overlap.time + hMdct->ov_size) || - (pOut1 >= output && pOut1 < output + 1024)); - *pOut1 += IMDCT_SCALE_DBL(-x1); - pOut1--; - } - - /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ - pOut1 += (fl / 2) + 1; - - pFAC_and_FAC_ZIR += fac_length; /* set pointer to beginning of FAC ZIR */ - - if (nl == 0) { - /* save pointer to write FAC ZIR data later */ - hMdct->pFacZir = pFAC_and_FAC_ZIR; - } else { - FDK_ASSERT(nl >= fac_length); - /* FAC ZIR will be added now ... */ - hMdct->pFacZir = NULL; - } - - pF = pFAC_and_FAC_ZIR; - f_len = fac_length; - - pCurr = pSpec + tl - fl / 2 - 1; - for (i = 0; i < nl; i++) { - FIXP_DBL x = -(*pCurr--); - /* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */ - if (i < f_len) { - x += *pF++; - } - - FDK_ASSERT((pOut1 >= hMdct->overlap.time && - pOut1 < hMdct->overlap.time + hMdct->ov_size) || - (pOut1 >= output && pOut1 < output + 1024)); - *pOut1 = IMDCT_SCALE_DBL(x); - pOut1++; - } - - hMdct->prev_nr = nr; - hMdct->prev_fr = fr; - hMdct->prev_wrs = wrs; - hMdct->prev_tl = tl; - hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; - hMdct->prevAliasSymmetry = currAliasingSymmetry; - fl = fr; - nl = nr; - - pOvl = pSpec + tl / 2 - 1; - pOut0 = pOut1; - - for (w = 1; w < nSpec; w++) /* for ACELP -> FD short */ - { - const FIXP_WTP *pWindow_prev; - - /* Setup window pointers */ - pWindow_prev = hMdct->prev_wrs; - - /* Current spectrum */ - pSpec = _pSpec + w * tl; - - scale = total_gain_e; - - /* For the second, third, etc. short frames the alisaing symmetry is equal, - * either (0,0) or (1,1) */ - if (currAliasingSymmetry == 0) { - /* DCT IV of current spectrum */ - dct_IV(pSpec, tl, &scale); - } else { - dst_IV(pSpec, tl, &scale); - } - - /* Optional scaling of time domain - no yet windowed - of current spectrum - */ - /* and de-scale current spectrum signal (time domain, no yet windowed) */ - if (total_gain != (FIXP_DBL)0) { - for (i = 0; i < tl; i++) { - pSpec[i] = fMult(pSpec[i], total_gain); - } - } - loc_scale = fixmin_I(spec_scale[w] + scale, (INT)DFRACT_BITS - 1); - scaleValuesSaturate(pSpec, tl, loc_scale); - - if (noOutSamples <= nrSamples) { - /* Divert output first half to overlap buffer if we already got enough - * output samples. */ - pOut0 = hMdct->overlap.time + hMdct->ov_offset; - hMdct->ov_offset += hMdct->prev_nr + fl / 2; - } else { - /* Account output samples */ - nrSamples += hMdct->prev_nr + fl / 2; - } - - /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */ - for (i = 0; i < hMdct->prev_nr; i++) { - FIXP_DBL x = -(*pOvl--); - *pOut0 = IMDCT_SCALE_DBL(x); - pOut0++; - } - - if (noOutSamples <= nrSamples) { - /* Divert output second half to overlap buffer if we already got enough - * output samples. */ - pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl / 2 - 1; - hMdct->ov_offset += fl / 2 + nl; - } else { - pOut1 = pOut0 + (fl - 1); - nrSamples += fl / 2 + nl; - } - - /* output samples before window crossing point NR .. TL/2. - * -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */ - /* output samples after window crossing point TL/2 .. TL/2+FL/2. - * -overlap[0..FL/2] - current[TL/2..FL/2] */ - pCurr = pSpec + tl - fl / 2; - if (currAliasingSymmetry == 0) { - for (i = 0; i < fl / 2; i++) { - FIXP_DBL x0, x1; - - cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(-x1); - pOut0++; - pOut1--; - } - } else { - if (hMdct->prevPrevAliasSymmetry == 0) { - /* Jump DST II -> DST IV for the second window */ - for (i = 0; i < fl / 2; i++) { - FIXP_DBL x0, x1; - - cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(x1); - pOut0++; - pOut1--; - } - } else { - /* Jump DST IV -> DST IV from the second window on */ - for (i = 0; i < fl / 2; i++) { - FIXP_DBL x0, x1; - - cplxMult(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]); - *pOut0 = IMDCT_SCALE_DBL(x0); - *pOut1 = IMDCT_SCALE_DBL(x1); - pOut0++; - pOut1--; - } - } - } - - if (hMdct->pFacZir != 0) { - /* add FAC ZIR of previous ACELP -> mdct transition */ - FIXP_DBL *pOut = pOut0 - fl / 2; - FDK_ASSERT(fl / 2 <= 128); - for (i = 0; i < fl / 2; i++) { - pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); - } - hMdct->pFacZir = NULL; - } - pOut0 += (fl / 2); - - /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ - pOut1 += (fl / 2) + 1; - pCurr = pSpec + tl - fl / 2 - 1; - for (i = 0; i < nl; i++) { - FIXP_DBL x = -(*pCurr--); - *pOut1 = IMDCT_SCALE_DBL(x); - pOut1++; - } - - /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */ - pOvl = pSpec + tl / 2 - 1; - - /* Previous window values. */ - hMdct->prev_nr = nr; - hMdct->prev_fr = fr; - hMdct->prev_tl = tl; - hMdct->prev_wrs = pWindow_prev; - hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; - hMdct->prevAliasSymmetry = currAliasingSymmetry; - } - - /* Save overlap */ - - pOvl = hMdct->overlap.freq + hMdct->ov_size - tl / 2; - FDK_ASSERT(pOvl >= hMdct->overlap.time + hMdct->ov_offset); - FDK_ASSERT(tl / 2 <= hMdct->ov_size); - for (i = 0; i < tl / 2; i++) { - pOvl[i] = _pSpec[i + (w - 1) * tl]; - } - - return nrSamples; -} diff --git a/libAACdec/src/usacdec_fac.h b/libAACdec/src/usacdec_fac.h deleted file mode 100644 index 100a6fa..0000000 --- a/libAACdec/src/usacdec_fac.h +++ /dev/null @@ -1,191 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Manuel Jander - - Description: USAC FAC - -*******************************************************************************/ - -#ifndef USACDEC_FAC_H -#define USACDEC_FAC_H - -#include "channelinfo.h" -#include "FDK_bitstream.h" - -/** - * \brief Get the address of a memory area of the spectral data memory were the - * FAC data can be stored into. - * \param spec SPECTRAL_PTR pointing to the current spectral data. - * \param mod the current LPD mod array. - * \param pState pointer to a private state variable which must be 0 for the - * first call and not changed externally. - * \param isFullbandLPD is 1 if fullband LPD mode is on, otherwise it is 0. - */ -FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo, - UCHAR mod[NB_SUBFR], int *pState); - -/** - * \brief read a fac bitstream data block. - * \param hBs a bit stream handle, where the fac bitstream data is located. - * \param pFac pointer to were the FAC data will be stored into. - * \param pFacScale pointer to were the FAC data scale value will be stored - * into. - * \param tcx_gain value to be used as FAC gain. If zero, read fac_gain from - * bitstream. - * \param tcx_gain_e exponen value of tcx_gain. - * \param frame the subframe to be considered from the current superframe. - * Always 0 for FD case. - * \return 0 on success, -1 on error. - */ -int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, SCHAR *pFacScale, - int length, int use_gain, int frame); - -/** - * \brief Apply TCX and ALFD gains to FAC data. - * \param fac_data pointer to FAC data. - * \param fac_length FAC length (128 or 96). - * \param tcx_gain TCX gain - * \param alfd_gains pointer to alfd gains. - * \param mod mod value (1,2,3) of TCX frame where the FAC signal needs to be - * applied. - */ -void CFac_ApplyGains(FIXP_DBL fac_data[LFAC], const INT fac_length, - const FIXP_DBL tcx_gain, const FIXP_DBL alfd_gains[], - const INT mod); - -/** - * \brief Do FAC transition from frequency domain to ACELP domain. - */ -INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac_data, - const int fac_data_e, FIXP_LPC *A, INT A_exp, - INT nrOutSamples, const INT fac_length, - const INT isFdFac, UCHAR prevWindowShape); - -/** - * \brief Do FAC transition from ACELP domain to frequency domain. - * \param hMdct MDCT context. - * \param output pointer for time domain output. - * \param pSpec pointer to MDCT spectrum input. - * \param spec_scale MDCT spectrum exponents. - * \param nSpec amount of contiguos MDCT spectra. - * \param pFac pointer to FAC MDCT domain data. - * \param fac_scale exponent of FAC data. - * \param fac_length length of FAC data. - * \param nrSamples room in samples in output buffer. - * \param tl MDCT transform length of pSpec. - * \param wrs right MDCT window slope. - * \param fr right MDCT window slope length. - * \param A LP domain filter coefficients. - * \param deemph_mem deemphasis filter state. - * \param gain gain to be applied to FAC data before overlap add. - * \param old_syn_mem Synthesis filter state. - * \param isFdFac indicates fac processing from or to FD. - * \param pFacData fac data stored for fullband LPD. - * \param elFlags element specific parser guidance flags. - * \param isFacForFullband indicates that fac is processed for fullband LPD. - */ -INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pSpec, - const SHORT spec_scale[], const int nSpec, - FIXP_DBL *pFac_data, const int fac_data_e, - const INT fac_length, INT nrSamples, const INT tl, - const FIXP_WTP *wrs, const INT fr, FIXP_LPC A[16], - INT A_exp, CAcelpStaticMem *acelp_mem, - const FIXP_DBL gain, const int last_frame_lost, - const int isFdFac, const UCHAR last_lpd, const int k, - int currAliasingSymmetry); - -#endif /* USACDEC_FAC_H */ diff --git a/libAACdec/src/usacdec_lpc.cpp b/libAACdec/src/usacdec_lpc.cpp deleted file mode 100644 index 271463f..0000000 --- a/libAACdec/src/usacdec_lpc.cpp +++ /dev/null @@ -1,1194 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand, Manuel Jander - - Description: USAC LPC/AVQ decode - -*******************************************************************************/ - -#include "usacdec_lpc.h" - -#include "usacdec_rom.h" -#include "FDK_trigFcts.h" - -#define NQ_MAX 36 - -/* - * Helper functions. - */ - -/** - * \brief Read unary code. - * \param hBs bitstream handle as data source. - * \return decoded value. - */ -static int get_vlclbf(HANDLE_FDK_BITSTREAM hBs) { - int result = 0; - - while (FDKreadBits(hBs, 1) && result <= NQ_MAX) { - result++; - } - return result; -} - -/** - * \brief Read bit count limited unary code. - * \param hBs bitstream handle as data source - * \param n max amount of bits to be read. - * \return decoded value. - */ -static int get_vlclbf_n(HANDLE_FDK_BITSTREAM hBs, int n) { - int result = 0; - - while (FDKreadBits(hBs, 1)) { - result++; - n--; - if (n <= 0) { - break; - } - } - - return result; -} - -/* - * Algebraic Vector Quantizer - */ - -/* ZF_SCALE must be greater than (number of FIXP_ZF)/2 - because the loss of precision caused by fPow2Div2 in RE8_PPV() */ -//#define ZF_SCALE ((NQ_MAX-3)>>1) -#define ZF_SCALE ((DFRACT_BITS / 2)) -#define FIXP_ZF FIXP_DBL -#define INT2ZF(x, s) (FIXP_ZF)((x) << (ZF_SCALE - (s))) -#define ZF2INT(x) (INT)((x) >> ZF_SCALE) - -/* 1.0 in ZF format format */ -#define ONEZF ((FIXP_ZF)INT2ZF(1, 0)) - -/* static */ -void nearest_neighbor_2D8(FIXP_ZF x[8], int y[8]) { - FIXP_ZF s, em, e[8]; - int i, j, sum; - - /* round x into 2Z^8 i.e. compute y=(y1,...,y8) such that yi = 2[xi/2] - where [.] is the nearest integer operator - in the mean time, compute sum = y1+...+y8 - */ - sum = 0; - for (i = 0; i < 8; i++) { - FIXP_ZF tmp; - /* round to ..., -2, 0, 2, ... ([-1..1[ --> 0) */ - if (x[i] < (FIXP_ZF)0) { - tmp = ONEZF - x[i]; - y[i] = -2 * ((ZF2INT(tmp)) >> 1); - } else { - tmp = ONEZF + x[i]; - y[i] = 2 * ((ZF2INT(tmp)) >> 1); - } - sum += y[i]; - } - /* check if y1+...+y8 is a multiple of 4 - if not, y is not round xj in the wrong way where j is defined by - j = arg max_i | xi -yi| - (this is called the Wagner rule) - */ - if (sum % 4) { - /* find j = arg max_i | xi -yi| */ - em = (FIXP_SGL)0; - j = 0; - for (i = 0; i < 8; i++) { - /* compute ei = xi-yi */ - e[i] = x[i] - INT2ZF(y[i], 0); - } - for (i = 0; i < 8; i++) { - /* compute |ei| = | xi-yi | */ - if (e[i] < (FIXP_ZF)0) { - s = -e[i]; - } else { - s = e[i]; - } - /* check if |ei| is maximal, if so, set j=i */ - if (em < s) { - em = s; - j = i; - } - } - /* round xj in the "wrong way" */ - if (e[j] < (FIXP_ZF)0) { - y[j] -= 2; - } else { - y[j] += 2; - } - } -} - -/*-------------------------------------------------------------- - RE8_PPV(x,y) - NEAREST NEIGHBOR SEARCH IN INFINITE LATTICE RE8 - the algorithm is based on the definition of RE8 as - RE8 = (2D8) U (2D8+[1,1,1,1,1,1,1,1]) - it applies the coset decoding of Sloane and Conway - (i) x: point in R^8 in 32-ZF_SCALE.ZF_SCALE format - (o) y: point in RE8 (8-dimensional integer vector) - -------------------------------------------------------------- -*/ -/* static */ -void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) { - int i, y0[8], y1[8]; - FIXP_ZF x1[8], tmp; - FIXP_DBL e; - - /* find the nearest neighbor y0 of x in 2D8 */ - nearest_neighbor_2D8(x, y0); - /* find the nearest neighbor y1 of x in 2D8+(1,...,1) (by coset decoding) */ - for (i = 0; i < 8; i++) { - x1[i] = x[i] - ONEZF; - } - nearest_neighbor_2D8(x1, y1); - for (i = 0; i < 8; i++) { - y1[i] += 1; - } - - /* compute e0=||x-y0||^2 and e1=||x-y1||^2 */ - e = (FIXP_DBL)0; - for (i = 0; i < 8; i++) { - tmp = x[i] - INT2ZF(y0[i], 0); - e += fPow2Div2( - tmp << r); /* shift left to ensure that no fract part bits get lost. */ - tmp = x[i] - INT2ZF(y1[i], 0); - e -= fPow2Div2(tmp << r); - } - /* select best candidate y0 or y1 to minimize distortion */ - if (e < (FIXP_DBL)0) { - for (i = 0; i < 8; i++) { - y[i] = y0[i]; - } - } else { - for (i = 0; i < 8; i++) { - y[i] = y1[i]; - } - } -} - -/* table look-up of unsigned value: find i where index >= table[i] - Note: range must be >= 2, index must be >= table[0] */ -static int table_lookup(const USHORT *table, unsigned int index, int range) { - int i; - - for (i = 4; i < range; i += 4) { - if (index < table[i]) { - break; - } - } - if (i > range) { - i = range; - } - - if (index < table[i - 2]) { - i -= 2; - } - if (index < table[i - 1]) { - i--; - } - i--; - - return (i); /* index >= table[i] */ -} - -/*-------------------------------------------------------------------------- - re8_decode_rank_of_permutation(rank, xs, x) - DECODING OF THE RANK OF THE PERMUTATION OF xs - (i) rank: index (rank) of a permutation - (i) xs: signed leader in RE8 (8-dimensional integer vector) - (o) x: point in RE8 (8-dimensional integer vector) - -------------------------------------------------------------------------- - */ -static void re8_decode_rank_of_permutation(int rank, int *xs, SHORT x[8]) { - INT a[8], w[8], B, fac, fac_B, target; - int i, j; - - /* --- pre-processing based on the signed leader xs --- - - compute the alphabet a=[a[0] ... a[q-1]] of x (q elements) - such that a[0]!=...!=a[q-1] - it is assumed that xs is sorted in the form of a signed leader - which can be summarized in 2 requirements: - a) |xs[0]| >= |xs[1]| >= |xs[2]| >= ... >= |xs[7]| - b) if |xs[i]|=|xs[i-1]|, xs[i]>=xs[i+1] - where |.| indicates the absolute value operator - - compute q (the number of symbols in the alphabet) - - compute w[0..q-1] where w[j] counts the number of occurences of - the symbol a[j] in xs - - compute B = prod_j=0..q-1 (w[j]!) where .! is the factorial */ - /* xs[i], xs[i-1] and ptr_w/a*/ - j = 0; - w[j] = 1; - a[j] = xs[0]; - B = 1; - for (i = 1; i < 8; i++) { - if (xs[i] != xs[i - 1]) { - j++; - w[j] = 1; - a[j] = xs[i]; - } else { - w[j]++; - B *= w[j]; - } - } - - /* --- actual rank decoding --- - the rank of x (where x is a permutation of xs) is based on - Schalkwijk's formula - it is given by rank=sum_{k=0..7} (A_k * fac_k/B_k) - the decoding of this rank is sequential and reconstructs x[0..7] - element by element from x[0] to x[7] - [the tricky part is the inference of A_k for each k...] - */ - - if (w[0] == 8) { - for (i = 0; i < 8; i++) { - x[i] = a[0]; /* avoid fac of 40320 */ - } - } else { - target = rank * B; - fac_B = 1; - /* decode x element by element */ - for (i = 0; i < 8; i++) { - fac = fac_B * fdk_dec_tab_factorial[i]; /* fac = 1..5040 */ - j = -1; - do { - target -= w[++j] * fac; - } while (target >= 0); /* max of 30 tests / SV */ - x[i] = a[j]; - /* update rank, denominator B (B_k) and counter w[j] */ - target += w[j] * fac; /* target = fac_B*B*rank */ - fac_B *= w[j]; - w[j]--; - } - } -} - -/*-------------------------------------------------------------------------- - re8_decode_base_index(n, I, y) - DECODING OF AN INDEX IN Qn (n=0,2,3 or 4) - (i) n: codebook number (*n is an integer defined in {0,2,3,4}) - (i) I: index of c (pointer to unsigned 16-bit word) - (o) y: point in RE8 (8-dimensional integer vector) - note: the index I is defined as a 32-bit word, but only - 16 bits are required (long can be replaced by unsigned integer) - -------------------------------------------------------------------------- - */ -static void re8_decode_base_index(int *n, UINT index, SHORT y[8]) { - int i, im, t, sign_code, ka, ks, rank, leader[8]; - - if (*n < 2) { - for (i = 0; i < 8; i++) { - y[i] = 0; - } - } else { - // index = (unsigned int)*I; - /* search for the identifier ka of the absolute leader (table-lookup) - Q2 is a subset of Q3 - the two cases are considered in the same branch - */ - switch (*n) { - case 2: - case 3: - i = table_lookup(fdk_dec_I3, index, NB_LDQ3); - ka = fdk_dec_A3[i]; - break; - case 4: - i = table_lookup(fdk_dec_I4, index, NB_LDQ4); - ka = fdk_dec_A4[i]; - break; - default: - FDK_ASSERT(0); - return; - } - /* reconstruct the absolute leader */ - for (i = 0; i < 8; i++) { - leader[i] = fdk_dec_Da[ka][i]; - } - /* search for the identifier ks of the signed leader (table look-up) - (this search is focused based on the identifier ka of the absolute - leader)*/ - t = fdk_dec_Ia[ka]; - im = fdk_dec_Ns[ka]; - ks = table_lookup(fdk_dec_Is + t, index, im); - - /* reconstruct the signed leader from its sign code */ - sign_code = 2 * fdk_dec_Ds[t + ks]; - for (i = 7; i >= 0; i--) { - leader[i] *= (1 - (sign_code & 2)); - sign_code >>= 1; - } - - /* compute and decode the rank of the permutation */ - rank = index - fdk_dec_Is[t + ks]; /* rank = index - cardinality offset */ - - re8_decode_rank_of_permutation(rank, leader, y); - } - return; -} - -/* re8_y2k(y,m,k) - VORONOI INDEXING (INDEX DECODING) k -> y - (i) k: Voronoi index k[0..7] - (i) m: Voronoi modulo (m = 2^r = 1<=2) - (i) r: Voronoi order (m = 2^r = 1<=2) - (o) y: 8-dimensional point y[0..7] in RE8 - */ -static void re8_k2y(int *k, int r, SHORT *y) { - int i, tmp, sum; - SHORT v[8]; - FIXP_ZF zf[8]; - - FDK_ASSERT(r <= ZF_SCALE); - - /* compute y = k M and z=(y-a)/m, where - M = [4 ] - [2 2 ] - [| \ ] - [2 2 ] - [1 1 _ 1 1] - a=(2,0,...,0) - m = 1<= 1; i--) { - tmp = 2 * k[i]; - sum += tmp; - y[i] += tmp; - zf[i] = INT2ZF(y[i], r); - } - y[0] += (4 * k[0] + sum); - zf[0] = INT2ZF(y[0] - 2, r); - /* find nearest neighbor v of z in infinite RE8 */ - RE8_PPV(zf, v, r); - /* compute y -= m v */ - for (i = 0; i < 8; i++) { - y[i] -= (SHORT)(v[i] << r); - } -} - -/*-------------------------------------------------------------------------- - RE8_dec(n, I, k, y) - MULTI-RATE INDEXING OF A POINT y in THE LATTICE RE8 (INDEX DECODING) - (i) n: codebook number (*n is an integer defined in {0,2,3,4,..,n_max}). n_max - = 36 (i) I: index of c (pointer to unsigned 16-bit word) (i) k: index of v - (8-dimensional vector of binary indices) = Voronoi index (o) y: point in RE8 - (8-dimensional integer vector) note: the index I is defined as a 32-bit word, - but only 16 bits are required (long can be replaced by unsigned integer) - - return 0 on success, -1 on error. - -------------------------------------------------------------------------- - */ -static int RE8_dec(int n, int I, int *k, FIXP_DBL *y) { - SHORT v[8]; - SHORT _y[8]; - UINT r; - int i; - - /* Check bound of codebook qn */ - if (n > NQ_MAX) { - return -1; - } - - /* decode the sub-indices I and kv[] according to the codebook number n: - if n=0,2,3,4, decode I (no Voronoi extension) - if n>4, Voronoi extension is used, decode I and kv[] */ - if (n <= 4) { - re8_decode_base_index(&n, I, _y); - for (i = 0; i < 8; i++) { - y[i] = (LONG)_y[i]; - } - } else { - /* compute the Voronoi modulo m = 2^r where r is extension order */ - r = ((n - 3) >> 1); - - while (n > 4) { - n -= 2; - } - /* decode base codebook index I into c (c is an element of Q3 or Q4) - [here c is stored in y to save memory] */ - re8_decode_base_index(&n, I, _y); - /* decode Voronoi index k[] into v */ - re8_k2y(k, r, v); - /* reconstruct y as y = m c + v (with m=2^r, r integer >=1) */ - for (i = 0; i < 8; i++) { - y[i] = (LONG)((_y[i] << r) + v[i]); - } - } - return 0; -} - -/**************************/ -/* start LPC decode stuff */ -/**************************/ -//#define M 16 -#define FREQ_MAX 6400.0f -#define FREQ_DIV 400.0f -#define LSF_GAP 50.0f - -/** - * \brief calculate inverse weighting factor and add non-weighted residual - * LSF vector to first stage LSF approximation - * \param lsfq first stage LSF approximation values. - * \param xq weighted residual LSF vector - * \param nk_mode code book number coding mode. - */ -static void lsf_weight_2st(FIXP_LPC *lsfq, FIXP_DBL *xq, int nk_mode) { - FIXP_LPC d[M_LP_FILTER_ORDER + 1]; - FIXP_SGL factor; - LONG w; /* inverse weight factor */ - int i; - - /* compute lsf distance */ - d[0] = lsfq[0]; - d[M_LP_FILTER_ORDER] = - FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - lsfq[M_LP_FILTER_ORDER - 1]; - for (i = 1; i < M_LP_FILTER_ORDER; i++) { - d[i] = lsfq[i] - lsfq[i - 1]; - } - - switch (nk_mode) { - case 0: - factor = FL2FXCONST_SGL(2.0f * 60.0f / FREQ_DIV); - break; /* abs */ - case 1: - factor = FL2FXCONST_SGL(2.0f * 65.0f / FREQ_DIV); - break; /* mid */ - case 2: - factor = FL2FXCONST_SGL(2.0f * 64.0f / FREQ_DIV); - break; /* rel1 */ - default: - factor = FL2FXCONST_SGL(2.0f * 63.0f / FREQ_DIV); - break; /* rel2 */ - } - /* add non-weighted residual LSF vector to LSF1st */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1]))); - lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)(w * (LONG)xq[i]))); - } - - return; -} - -/** - * \brief decode nqn amount of code book numbers. These values determine the - * amount of following bits for nqn AVQ RE8 vectors. - * \param nk_mode quantization mode. - * \param nqn amount code book number to read. - * \param qn pointer to output buffer to hold decoded code book numbers qn. - */ -static void decode_qn(HANDLE_FDK_BITSTREAM hBs, int nk_mode, int nqn, - int qn[]) { - int n; - - if (nk_mode == 1) { /* nk mode 1 */ - /* Unary code for mid LPC1/LPC3 */ - /* Q0=0, Q2=10, Q3=110, ... */ - for (n = 0; n < nqn; n++) { - qn[n] = get_vlclbf(hBs); - if (qn[n] > 0) { - qn[n]++; - } - } - } else { /* nk_mode 0, 3 and 2 */ - /* 2 bits to specify Q2,Q3,Q4,ext */ - for (n = 0; n < nqn; n++) { - qn[n] = 2 + FDKreadBits(hBs, 2); - } - if (nk_mode == 2) { - /* Unary code for rel LPC1/LPC3 */ - /* Q0 = 0, Q5=10, Q6=110, ... */ - for (n = 0; n < nqn; n++) { - if (qn[n] > 4) { - qn[n] = get_vlclbf(hBs); - if (qn[n] > 0) qn[n] += 4; - } - } - } else { /* nk_mode == (0 and 3) */ - /* Unary code for abs and rel LPC0/LPC2 */ - /* Q5 = 0, Q6=10, Q0=110, Q7=1110, ... */ - for (n = 0; n < nqn; n++) { - if (qn[n] > 4) { - qn[n] = get_vlclbf(hBs); - switch (qn[n]) { - case 0: - qn[n] = 5; - break; - case 1: - qn[n] = 6; - break; - case 2: - qn[n] = 0; - break; - default: - qn[n] += 4; - break; - } - } - } - } - } -} - -/** - * \brief reorder LSF coefficients to minimum distance. - * \param lsf pointer to buffer containing LSF coefficients and where reordered - * LSF coefficients will be stored into, scaled by LSF_SCALE. - * \param min_dist min distance scaled by LSF_SCALE - * \param n number of LSF/LSP coefficients. - */ -static void reorder_lsf(FIXP_LPC *lsf, FIXP_LPC min_dist, int n) { - FIXP_LPC lsf_min; - int i; - - lsf_min = min_dist; - for (i = 0; i < n; i++) { - if (lsf[i] < lsf_min) { - lsf[i] = lsf_min; - } - lsf_min = fAddSaturate(lsf[i], min_dist); - } - - /* reverse */ - lsf_min = FL2FXCONST_LPC(FREQ_MAX / (1 << LSF_SCALE)) - min_dist; - for (i = n - 1; i >= 0; i--) { - if (lsf[i] > lsf_min) { - lsf[i] = lsf_min; - } - - lsf_min = lsf[i] - min_dist; - } -} - -/** - * \brief First stage approximation - * \param hBs bitstream handle as data source - * \param lsfq pointer to output buffer to hold LPC coefficients scaled by - * LSF_SCALE. - */ -static void vlpc_1st_dec( - HANDLE_FDK_BITSTREAM hBs, /* input: codebook index */ - FIXP_LPC *lsfq /* i/o: i:prediction o:quantized lsf */ -) { - const FIXP_LPC *p_dico; - int i, index; - - index = FDKreadBits(hBs, 8); - p_dico = &fdk_dec_dico_lsf_abs_8b[index * M_LP_FILTER_ORDER]; - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsfq[i] = p_dico[i]; - } -} - -/** - * \brief Do first stage approximation weighting and multiply with AVQ - * refinement. - * \param hBs bitstream handle data ssource. - * \param lsfq buffer holding 1st stage approx, 2nd stage approx is added to - * this values. - * \param nk_mode quantization mode. - * \return 0 on success, -1 on error. - */ -static int vlpc_2st_dec( - HANDLE_FDK_BITSTREAM hBs, - FIXP_LPC *lsfq, /* i/o: i:1st stage o:1st+2nd stage */ - int nk_mode /* input: 0=abs, >0=rel */ -) { - int err; - FIXP_DBL xq[M_LP_FILTER_ORDER]; /* weighted residual LSF vector */ - - /* Decode AVQ refinement */ - { err = CLpc_DecodeAVQ(hBs, xq, nk_mode, 2, 8); } - if (err != 0) { - return -1; - } - - /* add non-weighted residual LSF vector to LSF1st */ - lsf_weight_2st(lsfq, xq, nk_mode); - - /* reorder */ - reorder_lsf(lsfq, FL2FXCONST_LPC(LSF_GAP / (1 << LSF_SCALE)), - M_LP_FILTER_ORDER); - - return 0; -} - -/* - * Externally visible functions - */ - -int CLpc_DecodeAVQ(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pOutput, int nk_mode, - int no_qn, int length) { - int i, l; - - for (i = 0; i < length; i += 8 * no_qn) { - int qn[2], nk, n, I; - int kv[8] = {0}; - - decode_qn(hBs, nk_mode, no_qn, qn); - - for (l = 0; l < no_qn; l++) { - if (qn[l] == 0) { - FDKmemclear(&pOutput[i + l * 8], 8 * sizeof(FIXP_DBL)); - } - - /* Voronoi extension order ( nk ) */ - nk = 0; - n = qn[l]; - if (qn[l] > 4) { - nk = (qn[l] - 3) >> 1; - n = qn[l] - nk * 2; - } - - /* Base codebook index, in reverse bit group order (!) */ - I = FDKreadBits(hBs, 4 * n); - - if (nk > 0) { - int j; - - for (j = 0; j < 8; j++) { - kv[j] = FDKreadBits(hBs, nk); - } - } - - if (RE8_dec(qn[l], I, kv, &pOutput[i + l * 8]) != 0) { - return -1; - } - } - } - return 0; -} - -int CLpc_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_LPC lsp[][M_LP_FILTER_ORDER], - FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], - FIXP_LPC lsf_adaptive_mean_cand[M_LP_FILTER_ORDER], - FIXP_SGL pStability[], UCHAR *mod, int first_lpd_flag, - int last_lpc_lost, int last_frame_ok) { - int i, k, err; - int mode_lpc_bin = 0; /* mode_lpc bitstream representation */ - int lpc_present[5] = {0, 0, 0, 0, 0}; - int lpc0_available = 1; - int s = 0; - int l = 3; - const int nbDiv = NB_DIV; - - lpc_present[4 >> s] = 1; /* LPC4 */ - - /* Decode LPC filters in the following order: LPC 4,0,2,1,3 */ - - /*** Decode LPC4 ***/ - vlpc_1st_dec(hBs, lsp[4 >> s]); - err = vlpc_2st_dec(hBs, lsp[4 >> s], 0); /* nk_mode = 0 */ - if (err != 0) { - return err; - } - - /*** Decode LPC0 and LPC2 ***/ - k = 0; - if (!first_lpd_flag) { - lpc_present[0] = 1; - lpc0_available = !last_lpc_lost; - /* old LPC4 is new LPC0 */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[0][i] = lpc4_lsf[i]; - } - /* skip LPC0 and continue with LPC2 */ - k = 2; - } - - for (; k < l; k += 2) { - int nk_mode = 0; - - if ((k == 2) && (mod[0] == 3)) { - break; /* skip LPC2 */ - } - - lpc_present[k >> s] = 1; - - mode_lpc_bin = FDKreadBit(hBs); - - if (mode_lpc_bin == 0) { - /* LPC0/LPC2: Abs */ - vlpc_1st_dec(hBs, lsp[k >> s]); - } else { - /* LPC0/LPC2: RelR */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[k >> s][i] = lsp[4 >> s][i]; - } - nk_mode = 3; - } - - err = vlpc_2st_dec(hBs, lsp[k >> s], nk_mode); - if (err != 0) { - return err; - } - } - - /*** Decode LPC1 ***/ - if (mod[0] < 2) { /* else: skip LPC1 */ - lpc_present[1] = 1; - mode_lpc_bin = get_vlclbf_n(hBs, 2); - - switch (mode_lpc_bin) { - case 1: - /* LPC1: abs */ - vlpc_1st_dec(hBs, lsp[1]); - err = vlpc_2st_dec(hBs, lsp[1], 0); - if (err != 0) { - return err; - } - break; - case 2: - /* LPC1: mid0 (no second stage AVQ quantizer in this case) */ - if (lpc0_available) { /* LPC0/lsf[0] might be zero some times */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[1][i] = (lsp[0][i] >> 1) + (lsp[2][i] >> 1); - } - } else { - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[1][i] = lsp[2][i]; - } - } - break; - case 0: - /* LPC1: RelR */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[1][i] = lsp[2][i]; - } - err = vlpc_2st_dec(hBs, lsp[1], 2 << s); - if (err != 0) { - return err; - } - break; - } - } - - /*** Decode LPC3 ***/ - if ((mod[2] < 2)) { /* else: skip LPC3 */ - int nk_mode = 0; - lpc_present[3] = 1; - - mode_lpc_bin = get_vlclbf_n(hBs, 3); - - switch (mode_lpc_bin) { - case 1: - /* LPC3: abs */ - vlpc_1st_dec(hBs, lsp[3]); - break; - case 0: - /* LPC3: mid */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[3][i] = (lsp[2][i] >> 1) + (lsp[4][i] >> 1); - } - nk_mode = 1; - break; - case 2: - /* LPC3: relL */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[3][i] = lsp[2][i]; - } - nk_mode = 2; - break; - case 3: - /* LPC3: relR */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[3][i] = lsp[4][i]; - } - nk_mode = 2; - break; - } - err = vlpc_2st_dec(hBs, lsp[3], nk_mode); - if (err != 0) { - return err; - } - } - - if (!lpc0_available && !last_frame_ok) { - /* LPC(0) was lost. Use next available LPC(k) instead */ - for (k = 1; k < (nbDiv + 1); k++) { - if (lpc_present[k]) { - for (i = 0; i < M_LP_FILTER_ORDER; i++) { -#define LSF_INIT_TILT (0.25f) - if (mod[0] > 0) { - lsp[0][i] = FX_DBL2FX_LPC( - fMult(lsp[k][i], FL2FXCONST_SGL(1.0f - LSF_INIT_TILT)) + - fMult(fdk_dec_lsf_init[i], FL2FXCONST_SGL(LSF_INIT_TILT))); - } else { - lsp[0][i] = lsp[k][i]; - } - } - break; - } - } - } - - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lpc4_lsf[i] = lsp[4 >> s][i]; - } - - { - FIXP_DBL divFac; - int last, numLpc = 0; - - i = nbDiv; - do { - numLpc += lpc_present[i--]; - } while (i >= 0 && numLpc < 3); - - last = i; - - switch (numLpc) { - case 3: - divFac = FL2FXCONST_DBL(1.0f / 3.0f); - break; - case 2: - divFac = FL2FXCONST_DBL(1.0f / 2.0f); - break; - default: - divFac = FL2FXCONST_DBL(1.0f); - break; - } - - /* get the adaptive mean for the next (bad) frame */ - for (k = 0; k < M_LP_FILTER_ORDER; k++) { - FIXP_DBL tmp = (FIXP_DBL)0; - for (i = nbDiv; i > last; i--) { - if (lpc_present[i]) { - tmp = fMultAdd(tmp >> 1, lsp[i][k], divFac); - } - } - lsf_adaptive_mean_cand[k] = FX_DBL2FX_LPC(tmp); - } - } - - /* calculate stability factor Theta. Needed for ACELP decoder and concealment - */ - { - FIXP_LPC *lsf_prev, *lsf_curr; - k = 0; - - FDK_ASSERT(lpc_present[0] == 1 && lpc_present[4 >> s] == 1); - lsf_prev = lsp[0]; - for (i = 1; i < (nbDiv + 1); i++) { - if (lpc_present[i]) { - FIXP_DBL tmp = (FIXP_DBL)0; - int j; - lsf_curr = lsp[i]; - - /* sum = tmp * 2^(LSF_SCALE*2 + 4) */ - for (j = 0; j < M_LP_FILTER_ORDER; j++) { - tmp += fPow2Div2((FIXP_SGL)(lsf_curr[j] - lsf_prev[j])) >> 3; - } - - /* tmp = (float)(FL2FXCONST_DBL(1.25f) - fMult(tmp, - * FL2FXCONST_DBL(1/400000.0f))); */ - tmp = FL2FXCONST_DBL(1.25f / (1 << LSF_SCALE)) - - fMult(tmp, FL2FXCONST_DBL((1 << (LSF_SCALE + 4)) / 400000.0f)); - if (tmp >= FL2FXCONST_DBL(1.0f / (1 << LSF_SCALE))) { - pStability[k] = FL2FXCONST_SGL(1.0f / 2.0f); - } else if (tmp < FL2FXCONST_DBL(0.0f)) { - pStability[k] = FL2FXCONST_SGL(0.0f); - } else { - pStability[k] = FX_DBL2FX_SGL(tmp << (LSF_SCALE - 1)); - } - - lsf_prev = lsf_curr; - k = i; - } else { - /* Mark stability value as undefined. */ - pStability[i] = (FIXP_SGL)-1; - } - } - } - - /* convert into LSP domain */ - for (i = 0; i < (nbDiv + 1); i++) { - if (lpc_present[i]) { - for (k = 0; k < M_LP_FILTER_ORDER; k++) { - lsp[i][k] = FX_DBL2FX_LPC( - fixp_cos(fMult(lsp[i][k], - FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), - LSF_SCALE - LSPARG_SCALE)); - } - } - } - - return 0; -} - -void CLpc_Conceal(FIXP_LPC lsp[][M_LP_FILTER_ORDER], - FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], - FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER], - const int first_lpd_flag) { - int i, j; - -#define BETA (FL2FXCONST_SGL(0.25f)) -#define ONE_BETA (FL2FXCONST_SGL(0.75f)) -#define BFI_FAC (FL2FXCONST_SGL(0.90f)) -#define ONE_BFI_FAC (FL2FXCONST_SGL(0.10f)) - - /* Frame loss concealment (could be improved) */ - - if (first_lpd_flag) { - /* Reset past LSF values */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[0][i] = lpc4_lsf[i] = fdk_dec_lsf_init[i]; - } - } else { - /* old LPC4 is new LPC0 */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[0][i] = lpc4_lsf[i]; - } - } - - /* LPC1 */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - FIXP_LPC lsf_mean = FX_DBL2FX_LPC(fMult(BETA, fdk_dec_lsf_init[i]) + - fMult(ONE_BETA, lsf_adaptive_mean[i])); - - lsp[1][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lpc4_lsf[i]) + - fMult(ONE_BFI_FAC, lsf_mean)); - } - - /* LPC2 - LPC4 */ - for (j = 2; j <= 4; j++) { - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - /* lsf_mean[i] = FX_DBL2FX_LPC(fMult((FIXP_LPC)(BETA + j * - FL2FXCONST_LPC(0.1f)), fdk_dec_lsf_init[i]) - + fMult((FIXP_LPC)(ONE_BETA - j * - FL2FXCONST_LPC(0.1f)), lsf_adaptive_mean[i])); */ - - FIXP_LPC lsf_mean = FX_DBL2FX_LPC( - fMult((FIXP_SGL)(BETA + (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))), - (FIXP_SGL)fdk_dec_lsf_init[i]) + - fMult( - (FIXP_SGL)(ONE_BETA - (FIXP_SGL)(j * (INT)FL2FXCONST_SGL(0.1f))), - lsf_adaptive_mean[i])); - - lsp[j][i] = FX_DBL2FX_LPC(fMult(BFI_FAC, lsp[j - 1][i]) + - fMult(ONE_BFI_FAC, lsf_mean)); - } - } - - /* Update past values for the future */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lpc4_lsf[i] = lsp[4][i]; - } - - /* convert into LSP domain */ - for (j = 0; j < 5; j++) { - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - lsp[j][i] = FX_DBL2FX_LPC(fixp_cos( - fMult(lsp[j][i], FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), - LSF_SCALE - LSPARG_SCALE)); - } - } -} - -void E_LPC_a_weight(FIXP_LPC *wA, const FIXP_LPC *A, int m) { - FIXP_DBL f; - int i; - - f = FL2FXCONST_DBL(0.92f); - for (i = 0; i < m; i++) { - wA[i] = FX_DBL2FX_LPC(fMult(A[i], f)); - f = fMult(f, FL2FXCONST_DBL(0.92f)); - } -} - -void CLpd_DecodeGain(FIXP_DBL *gain, INT *gain_e, int gain_code) { - /* gain * 2^(gain_e) = 10^(gain_code/28) */ - *gain = fLdPow( - FL2FXCONST_DBL(3.3219280948873623478703194294894 / 4.0), /* log2(10)*/ - 2, - fMultDiv2((FIXP_DBL)gain_code << (DFRACT_BITS - 1 - 7), - FL2FXCONST_DBL(2.0f / 28.0f)), - 7, gain_e); -} - - /** - * \brief * Find the polynomial F1(z) or F2(z) from the LSPs. - * This is performed by expanding the product polynomials: - * - * F1(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 ) - * i=0,2,4,6,8 - * F2(z) = product ( 1 - 2 LSP_i z^-1 + z^-2 ) - * i=1,3,5,7,9 - * - * where LSP_i are the LSPs in the cosine domain. - * R.A.Salami October 1990 - * \param lsp input, line spectral freq. (cosine domain) - * \param f output, the coefficients of F1 or F2, scaled by 8 bits - * \param n no of coefficients (m/2) - * \param flag 1 : F1(z) ; 2 : F2(z) - */ - -#define SF_F 8 - -static void get_lsppol(FIXP_LPC lsp[], FIXP_DBL f[], int n, int flag) { - FIXP_DBL b; - FIXP_LPC *plsp; - int i, j; - - plsp = lsp + flag - 1; - f[0] = FL2FXCONST_DBL(1.0f / (1 << SF_F)); - b = -FX_LPC2FX_DBL(*plsp); - f[1] = b >> (SF_F - 1); - for (i = 2; i <= n; i++) { - plsp += 2; - b = -FX_LPC2FX_DBL(*plsp); - f[i] = ((fMultDiv2(b, f[i - 1]) << 1) + (f[i - 2])) << 1; - for (j = i - 1; j > 1; j--) { - f[j] = f[j] + (fMultDiv2(b, f[j - 1]) << 2) + f[j - 2]; - } - f[1] = f[1] + (b >> (SF_F - 1)); - } - return; -} - -#define NC M_LP_FILTER_ORDER / 2 - -/** - * \brief lsp input LSP vector - * \brief a output LP filter coefficient vector scaled by SF_A_COEFFS. - */ -void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) { - FIXP_DBL f1[NC + 1], f2[NC + 1]; - int i, k; - - /*-----------------------------------------------------* - * Find the polynomials F1(z) and F2(z) * - *-----------------------------------------------------*/ - - get_lsppol(lsp, f1, NC, 1); - get_lsppol(lsp, f2, NC, 2); - - /*-----------------------------------------------------* - * Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1) * - *-----------------------------------------------------*/ - for (i = NC; i > 0; i--) { - f1[i] += f1[i - 1]; - f2[i] -= f2[i - 1]; - } - - FIXP_DBL aDBL[M_LP_FILTER_ORDER]; - - for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) { - FIXP_DBL tmp1, tmp2; - - tmp1 = f1[i] >> 1; - tmp2 = f2[i] >> 1; - - aDBL[i - 1] = (tmp1 + tmp2); - aDBL[k] = (tmp1 - tmp2); - } - - int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER); - - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a); - } - - *a_exp = 8 - headroom_a; -} diff --git a/libAACdec/src/usacdec_lpc.h b/libAACdec/src/usacdec_lpc.h deleted file mode 100644 index a6713c1..0000000 --- a/libAACdec/src/usacdec_lpc.h +++ /dev/null @@ -1,190 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Matthias Hildenbrand, Manuel Jander - - Description: USAC LPC/AVQ decode - -*******************************************************************************/ - -#ifndef USACDEC_LPC_H -#define USACDEC_LPC_H - -#include "channelinfo.h" -#include "common_fix.h" -#include "FDK_bitstream.h" -#include "usacdec_rom.h" - -#define LSPARG_SCALE 10 - -/** - * \brief AVQ (refinement) decode - * \param hBs bitstream handle - * \param lsfq buffer for AVQ decode output. - * \param nk_mode quantization mode. - * \param nqn amount of split/interleaved RE8 vectors. - * \param total amount of individual data values to decode. - * \return 0 on success, -1 on error. - */ -int CLpc_DecodeAVQ(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *lsfq, int nk_mode, - int nqn, int length); - -/** - * \brief Read and decode LPC coeficient sets. First stage approximation + AVQ - * decode. - * \param[in] hBs bitstream handle to read data from. - * \param[out] lsp buffer into which the decoded LSP coefficients will be stored - * into. - * \param[in,out] lpc4_lsf buffer into which the decoded LCP4 LSF coefficients - * will be stored into (persistent). - * \param[out] lsf_adaptive_mean_cand lsf adaptive mean vector needed for - * concealment. - * \param[out] pStability array with stability values for the ACELP decoder (and - * concealment). - * \param[in] mod array which defines modes (ACELP, TCX20|40|80) are used in - * the current superframe. - * \param[in] first_lpd_flag indicates the presence of LPC0 - * \param[in] last_lpc_lost indicate that LPC4 of previous frame was lost. - * \param[in] last_frame_ok indicate that the last frame was ok. - * \return 0 on success, -1 on error. - */ -int CLpc_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_LPC lsp[][M_LP_FILTER_ORDER], - FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], - FIXP_LPC lsf_adaptive_mean_cand[M_LP_FILTER_ORDER], - FIXP_SGL pStability[], UCHAR *mod, int first_lpd_flag, - int last_lpc_lost, int last_frame_ok); - -/** - * \brief Generate LPC coefficient sets in case frame loss. - * \param lsp buffer into which the decoded LSP coefficients will be stored - * into. - * \param lpc4_lsf buffer into which the decoded LCP4 LSF coefficients will be - * stored into (persistent). - * \param isf_adaptive_mean - * \param first_lpd_flag indicates the previous LSF4 coefficients lpc4_lsf[] are - * not valid. - */ -void CLpc_Conceal(FIXP_LPC lsp[][M_LP_FILTER_ORDER], - FIXP_LPC lpc4_lsf[M_LP_FILTER_ORDER], - FIXP_LPC isf_adaptive_mean[M_LP_FILTER_ORDER], - const int first_lpd_flag); - -/** - * \brief apply absolute weighting - * \param A weighted LPC coefficient vector output. The first coeffcient is - * implicitly 1.0 - * \param A LPC coefficient vector. The first coeffcient is implicitly 1.0 - * \param m length of vector A - */ -/* static */ -void E_LPC_a_weight(FIXP_LPC *wA, const FIXP_LPC *A, const int m); - -/** - * \brief decode TCX/FAC gain. In case of TCX the lg/sqrt(rms) part - * must still be applied to obtain the gain value. - * \param gain (o) pointer were the gain mantissa is stored into. - * \param gain_e (o) pointer were the gain exponent is stored into. - * \param gain_code (i) the 7 bit binary word from the bitstream - * representing the gain. - */ -void CLpd_DecodeGain(FIXP_DBL *gain, INT *gain_e, int gain_code); - -/** - * \brief convert LSP coefficients into LP domain. - */ -void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp); - -#endif /* USACDEC_LPC_H */ diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp deleted file mode 100644 index fcf7a76..0000000 --- a/libAACdec/src/usacdec_lpd.cpp +++ /dev/null @@ -1,2017 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Manuel Jander - - Description: USAC Linear Prediction Domain coding - -*******************************************************************************/ - -#include "usacdec_lpd.h" - -#include "usacdec_rom.h" -#include "usacdec_fac.h" -#include "usacdec_lpc.h" -#include "FDK_tools_rom.h" -#include "fft.h" -#include "mdct.h" -#include "usacdec_acelp.h" -#include "overlapadd.h" - -#include "block.h" - -#define SF_PITCH_TRACK 6 -#define SF_GAIN 3 -#define MIN_VAL FL2FXCONST_DBL(0.0f) -#define MAX_VAL (FIXP_DBL) MAXVAL_DBL - -#include "ac_arith_coder.h" - -void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, - const FIXP_SGL *filt, INT stop, int len) { - INT i, j; - FIXP_DBL tmp; - - for (i = 0; i < stop; i++) { - tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 - for (j = 1; j <= len; j++) { - tmp += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]); - } - syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp)); - } -} - -void bass_pf_1sf_delay( - FIXP_DBL *syn, /* (i) : 12.8kHz synthesis to postfilter */ - const INT *T_sf, /* (i) : Pitch period for all subframes (T_sf[16]) */ - FIXP_DBL *pit_gain, - const int frame_length, /* (i) : frame length (should be 768|1024) */ - const INT l_frame, - const INT l_next, /* (i) : look ahead for symmetric filtering */ - FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ - FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */ -{ - INT i, sf, i_subfr, T, T2, lg; - - FIXP_DBL tmp, ener, corr, gain; - FIXP_DBL *noise, *noise_in; - FIXP_DBL - noise_buf[L_FILT + (2 * L_SUBFR)]; // L_FILT = 12, L_SUBFR = 64 => 140 - const FIXP_DBL *x, *y; - - { - noise = noise_buf + L_FILT; // L_FILT = 12 delay of upsampling filter - noise_in = noise_buf + L_FILT + L_SUBFR; - /* Input scaling of the BPF memory */ - scaleValues(mem_bpf, (L_FILT + L_SUBFR), 1); - } - - int gain_exp = 17; - - sf = 0; - for (i_subfr = 0; i_subfr < l_frame; i_subfr += L_SUBFR, sf++) { - T = T_sf[sf]; - gain = pit_gain[sf]; - - /* Gain is in Q17.14 */ - /* If gain > 1 set to 1 */ - if (gain > (FIXP_DBL)(1 << 14)) gain = (FIXP_DBL)(1 << 14); - - /* If gain < 0 set to 0 */ - if (gain < (FIXP_DBL)0) gain = (FIXP_DBL)0; - - if (gain > (FIXP_DBL)0) { - /* pitch tracker: test pitch/2 to avoid continuous pitch doubling */ - /* Note: pitch is limited to PIT_MIN (34 = 376Hz) at the encoder */ - T2 = T >> 1; - x = &syn[i_subfr - L_EXTRA]; - y = &syn[i_subfr - T2 - L_EXTRA]; - - ener = (FIXP_DBL)0; - corr = (FIXP_DBL)0; - tmp = (FIXP_DBL)0; - - int headroom_x = getScalefactor(x, L_SUBFR + L_EXTRA); - int headroom_y = getScalefactor(y, L_SUBFR + L_EXTRA); - - int width_shift = 7; - - for (i = 0; i < (L_SUBFR + L_EXTRA); i++) { - ener += fPow2Div2((x[i] << headroom_x)) >> width_shift; - corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> - width_shift; - tmp += fPow2Div2((y[i] << headroom_y)) >> width_shift; - } - - int exp_ener = ((17 - headroom_x) << 1) + width_shift + 1; - int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; - int exp_tmp = ((17 - headroom_y) << 1) + width_shift + 1; - - /* Add 0.01 to "ener". Adjust exponents */ - FIXP_DBL point_zero_one = (FIXP_DBL)0x51eb851f; /* In Q-6.37 */ - int diff; - ener = fAddNorm(ener, exp_ener, point_zero_one, -6, &exp_ener); - corr = fAddNorm(corr, exp_corr, point_zero_one, -6, &exp_corr); - tmp = fAddNorm(tmp, exp_tmp, point_zero_one, -6, &exp_tmp); - - /* use T2 if normalized correlation > 0.95 */ - INT s1, s2; - s1 = CntLeadingZeros(ener) - 1; - s2 = CntLeadingZeros(tmp) - 1; - - FIXP_DBL ener_by_tmp = fMultDiv2(ener << s1, tmp << s2); - int ener_by_tmp_exp = (exp_ener - s1) + (exp_tmp - s2) + 1; - - if (ener_by_tmp_exp & 1) { - ener_by_tmp <<= 1; - ener_by_tmp_exp -= 1; - } - - int temp_exp = 0; - - FIXP_DBL temp1 = invSqrtNorm2(ener_by_tmp, &temp_exp); - - int temp1_exp = temp_exp - (ener_by_tmp_exp >> 1); - - FIXP_DBL tmp_result = fMult(corr, temp1); - - int tmp_result_exp = exp_corr + temp1_exp; - - diff = tmp_result_exp - 0; - FIXP_DBL point95 = FL2FXCONST_DBL(0.95f); - if (diff >= 0) { - diff = fMin(diff, 31); - point95 = FL2FXCONST_DBL(0.95f) >> diff; - } else { - diff = fMax(diff, -31); - tmp_result >>= (-diff); - } - - if (tmp_result > point95) T = T2; - - /* prevent that noise calculation below reaches into not defined signal - parts at the end of the synth_buf or in other words restrict the below - used index (i+i_subfr+T) < l_frame + l_next - */ - lg = l_frame + l_next - T - i_subfr; - - if (lg > L_SUBFR) - lg = L_SUBFR; - else if (lg < 0) - lg = 0; - - /* limit gain to avoid problem on burst */ - if (lg > 0) { - FIXP_DBL tmp1; - - /* max(lg) = 64 => scale with 6 bits minus 1 (fPow2Div2) */ - - s1 = getScalefactor(&syn[i_subfr], lg); - s2 = getScalefactor(&syn[i_subfr + T], lg); - INT s = fixMin(s1, s2); - - tmp = (FIXP_DBL)0; - ener = (FIXP_DBL)0; - for (i = 0; i < lg; i++) { - tmp += fPow2Div2(syn[i + i_subfr] << s1) >> (SF_PITCH_TRACK); - ener += fPow2Div2(syn[i + i_subfr + T] << s2) >> (SF_PITCH_TRACK); - } - tmp = tmp >> fMin(DFRACT_BITS - 1, (2 * (s1 - s))); - ener = ener >> fMin(DFRACT_BITS - 1, (2 * (s2 - s))); - - /* error robustness: for the specific case syn[...] == -1.0f for all 64 - samples ener or tmp might overflow and become negative. For all sane - cases we have enough headroom. - */ - if (ener <= (FIXP_DBL)0) { - ener = (FIXP_DBL)1; - } - if (tmp <= (FIXP_DBL)0) { - tmp = (FIXP_DBL)1; - } - FDK_ASSERT(ener > (FIXP_DBL)0); - - /* tmp = sqrt(tmp/ener) */ - int result_e = 0; - tmp1 = fDivNorm(tmp, ener, &result_e); - if (result_e & 1) { - tmp1 >>= 1; - result_e += 1; - } - tmp = sqrtFixp(tmp1); - result_e >>= 1; - - gain_exp = 17; - - diff = result_e - gain_exp; - - FIXP_DBL gain1 = gain; - - if (diff >= 0) { - diff = fMin(diff, 31); - gain1 >>= diff; - } else { - result_e += (-diff); - diff = fMax(diff, -31); - tmp >>= (-diff); - } - - if (tmp < gain1) { - gain = tmp; - gain_exp = result_e; - } - } - - /* calculate noise based on voiced pitch */ - /* fMultDiv2() replaces weighting of gain with 0.5 */ - diff = gain_exp - 17; - if (diff >= 0) { - gain <<= diff; - } else { - gain >>= (-diff); - } - - s1 = CntLeadingZeros(gain) - 1; - s1 -= 16; /* Leading bits for SGL */ - - FIXP_SGL gainSGL = FX_DBL2FX_SGL(gain << 16); - - gainSGL = gainSGL << s1; - - { - for (i = 0; i < lg; i++) { - /* scaled with SF_SYNTH + gain_sf + 1 */ - noise_in[i] = - (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) - - (syn[i + i_subfr + T] >> 1))) >> - s1; - } - - for (i = lg; i < L_SUBFR; i++) { - /* scaled with SF_SYNTH + gain_sf + 1 */ - noise_in[i] = - (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1; - } - } - } else { - FDKmemset(noise_in, (FIXP_DBL)0, L_SUBFR * sizeof(FIXP_DBL)); - } - - { - FDKmemcpy(noise_buf, mem_bpf, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); - - FDKmemcpy(mem_bpf, noise_buf + L_SUBFR, - (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); - } - - /* substract from voiced speech low-pass filtered noise */ - /* filter coefficients are scaled with factor SF_FILT_LP (1) */ - - { - filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise, - fdk_dec_filt_lp, L_SUBFR, L_FILT); - } - } - - { - - } - - // To be determined (info from Ben) - { - /* Output scaling of the BPF memory */ - scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1); - /* Copy the rest of the signal (after the fac) */ - scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame], - (FIXP_DBL *)&syn[l_frame - L_SUBFR], - (frame_length - l_frame), MDCT_OUT_HEADROOM); - } - - return; -} - -/* - * Frequency Domain Noise Shaping - */ - -/** - * \brief Adaptive Low Frequencies Deemphasis of spectral coefficients. - * - * Ensure quantization of low frequencies in case where the - * signal dynamic is higher than the LPC noise shaping. - * This is the inverse operation of adap_low_freq_emph(). - * Output gain of all blocks. - * - * \param x pointer to the spectral coefficients, requires 1 bit headroom. - * \param lg length of x. - * \param bUseNewAlfe if set, apply ALFD for fullband lpd. - * \param gainLpc1 pointer to gain based on old input LPC coefficients. - * \param gainLpc2 pointer to gain based on new input LPC coefficients. - * \param alfd_gains pointer to output gains. - * \param s current scale shift factor of x. - */ -#define ALFDPOW2_SCALE 3 -/*static*/ -void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[], - INT s) { - { - int i, j, k, i_max; - FIXP_DBL max, fac; - /* Note: This stack array saves temporary accumulation results to be used in - * a second run */ - /* The size should be limited to (1024/4)/8=32 */ - FIXP_DBL tmp_pow2[32]; - - s = s * 2 + ALFDPOW2_SCALE; - s = fMin(31, s); - - k = 8; - i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */ - - /* find spectral peak */ - max = FL2FX_DBL(0.01f) >> s; - for (i = 0; i < i_max; i += k) { - FIXP_DBL tmp; - - tmp = FIXP_DBL(0); - FIXP_DBL *pX = &x[i]; - - j = 8; - do { - FIXP_DBL x0 = *pX++; - FIXP_DBL x1 = *pX++; - x0 = fPow2Div2(x0); - x1 = fPow2Div2(x1); - tmp = tmp + (x0 >> (ALFDPOW2_SCALE - 1)); - tmp = tmp + (x1 >> (ALFDPOW2_SCALE - 1)); - } while ((j = j - 2) != 0); - tmp = fMax(tmp, (FL2FX_DBL(0.01f) >> s)); - tmp_pow2[i >> 3] = tmp; - if (tmp > max) { - max = tmp; - } - } - - /* deemphasis of all blocks below the peak */ - fac = FL2FX_DBL(0.1f) >> 1; - for (i = 0; i < i_max; i += k) { - FIXP_DBL tmp; - INT shifti; - - tmp = tmp_pow2[i >> 3]; - - /* tmp = (float)sqrt(tmp/max); */ - - /* value of tmp is between 8/2*max^2 and max^2 / 2. */ - /* required shift factor of division can grow up to 27 - (grows exponentially for values toward zero) - thus using normalized division to assure valid result. */ - { - INT sd; - - if (tmp != (FIXP_DBL)0) { - tmp = fDivNorm(max, tmp, &sd); - if (sd & 1) { - sd++; - tmp >>= 1; - } - } else { - tmp = (FIXP_DBL)MAXVAL_DBL; - sd = 0; - } - tmp = invSqrtNorm2(tmp, &shifti); - tmp = scaleValue(tmp, shifti - 1 - (sd / 2)); - } - if (tmp > fac) { - fac = tmp; - } - FIXP_DBL *pX = &x[i]; - - j = 8; - do { - FIXP_DBL x0 = pX[0]; - FIXP_DBL x1 = pX[1]; - x0 = fMultDiv2(x0, fac); - x1 = fMultDiv2(x1, fac); - x0 = x0 << 2; - x1 = x1 << 2; - *pX++ = x0; - *pX++ = x1; - - } while ((j = j - 2) != 0); - /* Store gains for FAC */ - *alfd_gains++ = fac; - } - } -} - -/** - * \brief Interpolated Noise Shaping for mdct coefficients. - * This algorithm shapes temporally the spectral noise between - * the two spectral noise represention (FDNS_NPTS of resolution). - * The noise is shaped monotonically between the two points - * using a curved shape to favor the lower gain in mid-frame. - * ODFT and amplitud calculation are applied to the 2 LPC coefficients first. - * - * \param r pointer to spectrum data. - * \param rms RMS of output spectrum. - * \param lg length of r. - * \param A1 pointer to old input LPC coefficients of length M_LP_FILTER_ORDER - * scaled by SF_A_COEFFS. - * \param A2 pointer to new input LPC coefficients of length M_LP_FILTER_ORDER - * scaled by SF_A_COEFFS. - * \param bLpc2Mdct flags control lpc2mdct conversion and noise shaping. - * \param gainLpc1 pointer to gain based on old input LPC coefficients. - * \param gainLpc2 pointer to gain based on new input LPC coefficients. - * \param gLpc_e pointer to exponent of gainLpc1 and gainLpc2. - */ -/* static */ -#define NSHAPE_SCALE (4) - -#define LPC2MDCT_CALC (1) -#define LPC2MDCT_GAIN_LOAD (2) -#define LPC2MDCT_GAIN_SAVE (4) -#define LPC2MDCT_APPLY_NSHAPE (8) - -void lpc2mdctAndNoiseShaping(FIXP_DBL *r, SHORT *pScale, const INT lg, - const INT fdns_npts, const FIXP_LPC *A1, - const INT A1_exp, const FIXP_LPC *A2, - const INT A2_exp) { - FIXP_DBL *tmp2 = NULL; - FIXP_DBL rr_minus_one; - int i, k, s, step; - - C_AALLOC_SCRATCH_START(tmp1, FIXP_DBL, FDNS_NPTS * 8) - - { - tmp2 = tmp1 + fdns_npts * 4; - - /* lpc2mdct() */ - - /* ODFT. E_LPC_a_weight() for A1 and A2 vectors is included into the loop - * below. */ - FIXP_DBL f = FL2FXCONST_DBL(0.92f); - - const FIXP_STP *SinTab; - int k_step; - /* needed values: sin(phi), cos(phi); phi = i*PI/(2*fdns_npts), i = 0 ... - * M_LP_FILTER_ORDER */ - switch (fdns_npts) { - case 64: - SinTab = SineTable512; - k_step = (512 / 64); - FDK_ASSERT(512 >= 64); - break; - case 48: - SinTab = SineTable384; - k_step = 384 / 48; - FDK_ASSERT(384 >= 48); - break; - default: - FDK_ASSERT(0); - return; - } - - for (i = 0, k = k_step; i < M_LP_FILTER_ORDER; i++, k += k_step) { - FIXP_STP cs = SinTab[k]; - FIXP_DBL wA1, wA2; - - wA1 = fMult(A1[i], f); - wA2 = fMult(A2[i], f); - - /* r[i] = A[i]*cos() */ - tmp1[2 + i * 2] = fMult(wA1, cs.v.re); - tmp2[2 + i * 2] = fMult(wA2, cs.v.re); - /* i[i] = A[i]*sin() */ - tmp1[3 + i * 2] = -fMult(wA1, cs.v.im); - tmp2[3 + i * 2] = -fMult(wA2, cs.v.im); - - f = fMult(f, FL2FXCONST_DBL(0.92f)); - } - - /* Guarantee at least 2 bits of headroom for the FFT */ - /* "3" stands for 1.0 with 2 bits of headroom; (A1_exp + 2) guarantess 2 - * bits of headroom if A1_exp > 1 */ - int A1_exp_fix = fMax(3, A1_exp + 2); - int A2_exp_fix = fMax(3, A2_exp + 2); - - /* Set 1.0 in the proper format */ - tmp1[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A1_exp_fix); - tmp2[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A2_exp_fix); - - tmp1[1] = tmp2[1] = (FIXP_DBL)0; - - /* Clear the resto of the array */ - FDKmemclear( - tmp1 + 2 * (M_LP_FILTER_ORDER + 1), - 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); - FDKmemclear( - tmp2 + 2 * (M_LP_FILTER_ORDER + 1), - 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); - - /* Guarantee 2 bits of headroom for FFT */ - scaleValues(&tmp1[2], (2 * M_LP_FILTER_ORDER), (A1_exp - A1_exp_fix)); - scaleValues(&tmp2[2], (2 * M_LP_FILTER_ORDER), (A2_exp - A2_exp_fix)); - - INT s2; - s = A1_exp_fix; - s2 = A2_exp_fix; - - fft(2 * fdns_npts, tmp1, &s); - fft(2 * fdns_npts, tmp2, &s2); - - /* Adjust the exponents of both fft outputs if necessary*/ - if (s > s2) { - scaleValues(tmp2, 2 * fdns_npts, s2 - s); - s2 = s; - } else if (s < s2) { - scaleValues(tmp1, 2 * fdns_npts, s - s2); - s = s2; - } - - FDK_ASSERT(s == s2); - } - - /* Get amplitude and apply gains */ - step = lg / fdns_npts; - rr_minus_one = (FIXP_DBL)0; - - for (k = 0; k < fdns_npts; k++) { - FIXP_DBL g1, g2, inv_g1_g2, a, b; - INT inv_g1_g2_e; - int g_e, shift; - - { - FIXP_DBL real, imag; - int si1, si2, sInput; - - real = tmp1[k * 2]; - imag = tmp1[k * 2 + 1]; - sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); - real <<= sInput; - imag <<= sInput; - /* g1_e = si1 - 2*s/2 */ - g1 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si1); - si1 += sInput; - - real = tmp2[k * 2]; - imag = tmp2[k * 2 + 1]; - sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); - real <<= sInput; - imag <<= sInput; - /* g2_e = si2 - 2*s/2 */ - g2 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si2); - si2 += sInput; - - /* Pick a common scale factor for g1 and g2 */ - if (si1 > si2) { - g2 >>= si1 - si2; - g_e = si1 - s; - } else { - g1 >>= si2 - si1; - g_e = si2 - s; - } - } - - /* end of lpc2mdct() */ - - FDK_ASSERT(g1 >= (FIXP_DBL)0); - FDK_ASSERT(g2 >= (FIXP_DBL)0); - - /* mdct_IntNoiseShaping() */ - { - /* inv_g1_g2 * 2^inv_g1_g2_e = 1/(g1+g2) */ - inv_g1_g2 = (g1 >> 1) + (g2 >> 1); - if (inv_g1_g2 != (FIXP_DBL)0) { - inv_g1_g2 = fDivNorm(FL2FXCONST_DBL(0.5f), inv_g1_g2, &inv_g1_g2_e); - inv_g1_g2_e = inv_g1_g2_e - g_e; - } else { - inv_g1_g2 = (FIXP_DBL)MAXVAL_DBL; - inv_g1_g2_e = 0; - } - - if (g_e < 0) { - /* a_e = g_e + inv_g1_g2_e + 1 */ - a = scaleValue(fMult(fMult(g1, g2), inv_g1_g2), g_e); - /* b_e = g_e + inv_g1_g2_e */ - b = fMult(g2 - g1, inv_g1_g2); - shift = g_e + inv_g1_g2_e + 1 - NSHAPE_SCALE; - } else { - /* a_e = (g_e+g_e) + inv_g1_g2_e + 1 */ - a = fMult(fMult(g1, g2), inv_g1_g2); - /* b_e = (g_e+g_e) + inv_g1_g2_e */ - b = scaleValue(fMult(g2 - g1, inv_g1_g2), -g_e); - shift = (g_e + g_e) + inv_g1_g2_e + 1 - NSHAPE_SCALE; - } - - for (i = k * step; i < (k + 1) * step; i++) { - FIXP_DBL tmp; - - /* rr[i] = 2*a*r[i] + b*rr[i-1] */ - tmp = fMult(a, r[i]); - tmp += scaleValue(fMultDiv2(b, rr_minus_one), NSHAPE_SCALE); - tmp = scaleValueSaturate(tmp, shift); - rr_minus_one = tmp; - r[i] = tmp; - } - } - } - - /* end of mdct_IntNoiseShaping() */ - { *pScale += NSHAPE_SCALE; } - - C_AALLOC_SCRATCH_END(tmp1, FIXP_DBL, FDNS_NPTS * 8) -} - -/** - * \brief Calculates the energy. - * \param r pointer to spectrum. - * \param rs scale factor of spectrum r. - * \param lg frame length in audio samples. - * \param rms_e pointer to exponent of energy value. - * \return mantissa of energy value. - */ -static FIXP_DBL calcEnergy(const FIXP_DBL *r, const SHORT rs, const INT lg, - INT *rms_e) { - int headroom = getScalefactor(r, lg); - - FIXP_DBL rms_m = 0; - - /* Calculate number of growth bits due to addition */ - INT shift = (INT)(fNormz((FIXP_DBL)lg)); - shift = 31 - shift; - - /* Generate 1e-2 in Q-6.37 */ - const FIXP_DBL value0_01 = 0x51eb851e; - const INT value0_01_exp = -6; - - /* Find the exponent of the resulting energy value */ - *rms_e = ((rs - headroom) << 1) + shift + 1; - - INT delta = *rms_e - value0_01_exp; - if (delta > 0) { - /* Limit shift_to 31*/ - delta = fMin(31, delta); - rms_m = value0_01 >> delta; - } else { - rms_m = value0_01; - *rms_e = value0_01_exp; - shift = shift - delta; - /* Limit shift_to 31*/ - shift = fMin(31, shift); - } - - for (int i = 0; i < lg; i++) { - rms_m += fPow2Div2(r[i] << headroom) >> shift; - } - - return rms_m; -} - -/** - * \brief TCX gain calculation. - * \param pAacDecoderChannelInfo channel context data. - * \param r output spectrum. - * \param rms_e pointer to mantissa of energy value. - * \param rms_e pointer to exponent of energy value. - * \param frame the frame index of the LPD super frame. - * \param lg the frame length in audio samples. - * \param gain_m pointer to mantissa of TCX gain. - * \param gain_e pointer to exponent of TCX gain. - * \param elFlags element specific parser guidance flags. - * \param lg_fb the fullband frame length in audio samples. - * \param IGF_bgn the IGF start index. - */ -static void calcTCXGain(CAacDecoderChannelInfo *pAacDecoderChannelInfo, - FIXP_DBL *r, FIXP_DBL rms_m, INT rms_e, const INT frame, - const INT lg) { - if ((rms_m != (FIXP_DBL)0)) { - FIXP_DBL tcx_gain_m; - INT tcx_gain_e; - - CLpd_DecodeGain(&tcx_gain_m, &tcx_gain_e, - pAacDecoderChannelInfo->pDynData->specificTo.usac - .tcx_global_gain[frame]); - - /* rms * 2^rms_e = lg/sqrt(sum(spec^2)) */ - if (rms_e & 1) { - rms_m >>= 1; - rms_e++; - } - - { - FIXP_DBL fx_lg; - INT fx_lg_e, s; - INT inv_e; - - /* lg = fx_lg * 2^fx_lg_e */ - s = fNorm((FIXP_DBL)lg); - fx_lg = (FIXP_DBL)lg << s; - fx_lg_e = DFRACT_BITS - 1 - s; - /* 1/sqrt(rms) */ - rms_m = invSqrtNorm2(rms_m, &inv_e); - rms_m = fMult(rms_m, fx_lg); - rms_e = inv_e - (rms_e >> 1) + fx_lg_e; - } - - { - int s = fNorm(tcx_gain_m); - tcx_gain_m = tcx_gain_m << s; - tcx_gain_e -= s; - } - - tcx_gain_m = fMultDiv2(tcx_gain_m, rms_m); - tcx_gain_e = tcx_gain_e + rms_e; - - /* global_gain * 2^(global_gain_e+rms_e) = (10^(global_gain/28)) * rms * - * 2^rms_e */ - { - { tcx_gain_e += 1; } - } - - pAacDecoderChannelInfo->data.usac.tcx_gain[frame] = tcx_gain_m; - pAacDecoderChannelInfo->data.usac.tcx_gain_e[frame] = tcx_gain_e; - - pAacDecoderChannelInfo->specScale[frame] += tcx_gain_e; - } -} - -/** - * \brief FDNS decoding. - * \param pAacDecoderChannelInfo channel context data. - * \param pAacDecoderStaticChannelInfo channel context static data. - * \param r output spectrum. - * \param lg the frame length in audio samples. - * \param frame the frame index of the LPD super frame. - * \param pScale pointer to current scale shift factor of r[]. - * \param A1 old input LPC coefficients of length M_LP_FILTER_ORDER. - * \param A2 new input LPC coefficients of length M_LP_FILTER_ORDER. - * \param pAlfdGains pointer for ALFD gains output scaled by 1. - * \param fdns_npts number of lines (FDNS_NPTS). - * \param inf_mask pointer to noise mask. - * \param IGF_win_mode IGF window mode (LONG, SHORT, TCX10, TCX20). - * \param frameType (IGF_FRAME_DIVISION_AAC_OR_TCX_LONG or - * IGF_FRAME_DIVISION_TCX_SHORT_1). - * \param elFlags element specific parser guidance flags. - * \param lg_fb the fullband frame length in audio samples. - * \param IGF_bgn the IGF start index. - * \param rms_m mantisse of energy. - * \param rms_e exponent of energy. - */ -/* static */ -void CLpd_FdnsDecode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - FIXP_DBL r[], const INT lg, const INT frame, SHORT *pScale, - const FIXP_LPC A1[M_LP_FILTER_ORDER], const INT A1_exp, - const FIXP_LPC A2[M_LP_FILTER_ORDER], const INT A2_exp, - FIXP_DBL pAlfdGains[LFAC / 4], const INT fdns_npts) { - /* Weight LPC coefficients using Rm values */ - CLpd_AdaptLowFreqDeemph(r, lg, pAlfdGains, *pScale); - - FIXP_DBL rms_m = (FIXP_DBL)0; - INT rms_e = 0; - { - /* Calculate Energy */ - rms_m = calcEnergy(r, *pScale, lg, &rms_e); - } - - calcTCXGain(pAacDecoderChannelInfo, r, rms_m, rms_e, frame, lg); - - /* Apply ODFT and Noise Shaping. LP coefficient (A1, A2) weighting is done - * inside on the fly. */ - - lpc2mdctAndNoiseShaping(r, pScale, lg, fdns_npts, A1, A1_exp, A2, A2_exp); -} - -/** - * find pitch for TCX20 (time domain) concealment. - */ -static int find_mpitch(FIXP_DBL xri[], int lg) { - FIXP_DBL max, pitch; - INT pitch_e; - int i, n; - - max = (FIXP_DBL)0; - n = 2; - - /* find maximum below 400Hz */ - for (i = 2; i < (lg >> 4); i += 2) { - FIXP_DBL tmp = fPow2Div2(xri[i]) + fPow2Div2(xri[i + 1]); - if (tmp > max) { - max = tmp; - n = i; - } - } - - // pitch = ((float)lg<<1)/(float)n; - pitch = fDivNorm((FIXP_DBL)lg << 1, (FIXP_DBL)n, &pitch_e); - pitch >>= fixMax(0, DFRACT_BITS - 1 - pitch_e - 16); - - /* find pitch multiple under 20ms */ - if (pitch >= (FIXP_DBL)((256 << 16) - 1)) { /*231.0f*/ - n = 256; - } else { - FIXP_DBL mpitch = pitch; - while (mpitch < (FIXP_DBL)(255 << 16)) { - mpitch += pitch; - } - n = (int)(mpitch - pitch) >> 16; - } - - return (n); -} - -/** - * number of spectral coefficients / time domain samples using frame mode as - * index. - */ -static const int lg_table_ccfl[2][4] = { - {256, 256, 512, 1024}, /* coreCoderFrameLength = 1024 */ - {192, 192, 384, 768} /* coreCoderFrameLength = 768 */ -}; - -/** - * \brief Decode and render one MDCT-TCX frame. - * \param pAacDecoderChannelInfo channel context data. - * \param lg the frame length in audio samples. - * \param frame the frame index of the LPD super frame. - */ -static void CLpd_TcxDecode( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags, - int mod, int last_mod, int frame, int frameOk) { - FIXP_DBL *pAlfd_gains = pAacDecoderStaticChannelInfo->last_alfd_gains; - ULONG *pSeed = &pAacDecoderStaticChannelInfo->nfRandomSeed; - int lg = (pAacDecoderChannelInfo->granuleLength == 128) - ? lg_table_ccfl[0][mod + 0] - : lg_table_ccfl[1][mod + 0]; - int next_frame = frame + (1 << (mod - 1)); - int isFullBandLpd = 0; - - /* Obtain r[] vector by combining the quant[] and noise[] vectors */ - { - FIXP_DBL noise_level; - FIXP_DBL *coeffs = - SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, - pAacDecoderChannelInfo->granuleLength, isFullBandLpd); - int scale = pAacDecoderChannelInfo->specScale[frame]; - int i, nfBgn, nfEnd; - UCHAR tcx_noise_factor = pAacDecoderChannelInfo->pDynData->specificTo.usac - .tcx_noise_factor[frame]; - - /* find pitch for bfi case */ - pAacDecoderStaticChannelInfo->last_tcx_pitch = find_mpitch(coeffs, lg); - - if (frameOk) { - /* store for concealment */ - pAacDecoderStaticChannelInfo->last_tcx_noise_factor = tcx_noise_factor; - } else { - /* restore last frames value */ - tcx_noise_factor = pAacDecoderStaticChannelInfo->last_tcx_noise_factor; - } - - noise_level = - (FIXP_DBL)((LONG)FL2FXCONST_DBL(0.0625f) * (8 - tcx_noise_factor)); - noise_level = scaleValue(noise_level, -scale); - - const FIXP_DBL neg_noise_level = -noise_level; - - { - nfBgn = lg / 6; - nfEnd = lg; - } - - for (i = nfBgn; i < nfEnd - 7; i += 8) { - LONG tmp; - - /* Fill all 8 consecutive zero coeffs with noise */ - tmp = coeffs[i + 0] | coeffs[i + 1] | coeffs[i + 2] | coeffs[i + 3] | - coeffs[i + 4] | coeffs[i + 5] | coeffs[i + 6] | coeffs[i + 7]; - - if (tmp == 0) { - for (int k = i; k < i + 8; k++) { - UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) - : (coeffs[k] = noise_level); - } - } - } - if ((nfEnd - i) > - 0) { /* noise filling for last "band" with less than 8 bins */ - LONG tmp = (LONG)coeffs[i]; - int k; - - FDK_ASSERT((nfEnd - i) < 8); - for (k = 1; k < (nfEnd - i); k++) { - tmp |= (LONG)coeffs[i + k]; - } - if (tmp == 0) { - for (k = i; k < nfEnd; k++) { - UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) - : (coeffs[k] = noise_level); - } - } - } - } - - { - /* Convert LPC to LP domain */ - if (last_mod == 0) { - /* Note: The case where last_mod == 255 is handled by other means - * in CLpdChannelStream_Read() */ - E_LPC_f_lsp_a_conversion( - pAacDecoderChannelInfo->data.usac.lsp_coeff[frame], - pAacDecoderChannelInfo->data.usac.lp_coeff[frame], - &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame]); - } - - E_LPC_f_lsp_a_conversion( - pAacDecoderChannelInfo->data.usac.lsp_coeff[next_frame], - pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], - &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame]); - - /* FDNS decoding */ - CLpd_FdnsDecode( - pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, - SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, - pAacDecoderChannelInfo->granuleLength, isFullBandLpd), - lg, frame, pAacDecoderChannelInfo->specScale + frame, - pAacDecoderChannelInfo->data.usac.lp_coeff[frame], - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame], - pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame], pAlfd_gains, - pAacDecoderChannelInfo->granuleLength / 2 /* == FDNS_NPTS(ccfl) */ - ); - } -} - -/** - * \brief Read the tcx_coding bitstream part - * \param hBs bitstream handle to read from. - * \param pAacDecoderChannelInfo channel context info to store data into. - * \param lg the frame length in audio samples. - * \param first_tcx_flag flag indicating that this is the first TCX frame. - * \param frame the frame index of the LPD super frame. - */ -static AAC_DECODER_ERROR CLpd_TCX_Read( - HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, int lg, - int first_tcx_flag, int frame, UINT flags) { - AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; - ARITH_CODING_ERROR error = ARITH_CODER_OK; - FIXP_DBL *pSpec; - int arith_reset_flag = 0; - int isFullBandLpd = 0; - - pSpec = SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, - pAacDecoderChannelInfo->granuleLength, isFullBandLpd); - - /* TCX noise level */ - { - pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_noise_factor[frame] = - FDKreadBits(hBs, 3); - } - /* TCX global gain */ - pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_global_gain[frame] = - FDKreadBits(hBs, 7); - - /* Arithmetic coded residual/spectrum */ - if (first_tcx_flag) { - if (flags & AC_INDEP) { - arith_reset_flag = 1; - } else { - arith_reset_flag = FDKreadBits(hBs, 1); - } - } - - /* CArco_DecodeArithData() output scale of "pSpec" is DFRACT_BITS-1 */ - error = CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, pSpec, - lg, lg, arith_reset_flag); - - /* Rescale residual/spectrum */ - { - int scale = getScalefactor(pSpec, lg) - 2; /* Leave 2 bits headroom */ - - /* Exponent of CArco_DecodeArithData() output is DFRACT_BITS; integer - * values. */ - scaleValues(pSpec, lg, scale); - scale = DFRACT_BITS - 1 - scale; - - pAacDecoderChannelInfo->specScale[frame] = scale; - } - - if (error == ARITH_CODER_ERROR) errorAAC = AAC_DEC_UNKNOWN; - - return errorAAC; -} - -/** - * \brief translate lpd_mode into the mod[] array which describes the mode of - * each each LPD frame - * \param mod[] the array that will be filled with the mode indexes of the - * inidividual frames. - * \param lpd_mode the lpd_mode field read from the lpd_channel_stream - */ -static AAC_DECODER_ERROR CLpd_ReadAndMapLpdModeToModArray( - UCHAR mod[4], HANDLE_FDK_BITSTREAM hBs, UINT elFlags) { - int lpd_mode; - - { - lpd_mode = FDKreadBits(hBs, 5); - - if (lpd_mode > 25 || lpd_mode < 0) { - return AAC_DEC_PARSE_ERROR; - } - - switch (lpd_mode) { - case 25: - /* 1 80MS frame */ - mod[0] = mod[1] = mod[2] = mod[3] = 3; - break; - case 24: - /* 2 40MS frames */ - mod[0] = mod[1] = mod[2] = mod[3] = 2; - break; - default: - switch (lpd_mode >> 2) { - case 4: - /* lpd_mode 19 - 16 => 1 40MS and 2 20MS frames */ - mod[0] = mod[1] = 2; - mod[2] = (lpd_mode & 1) ? 1 : 0; - mod[3] = (lpd_mode & 2) ? 1 : 0; - break; - case 5: - /* lpd_mode 23 - 20 => 2 20MS and 1 40MS frames */ - mod[2] = mod[3] = 2; - mod[0] = (lpd_mode & 1) ? 1 : 0; - mod[1] = (lpd_mode & 2) ? 1 : 0; - break; - default: - /* lpd_mode < 16 => 4 20MS frames */ - mod[0] = (lpd_mode & 1) ? 1 : 0; - mod[1] = (lpd_mode & 2) ? 1 : 0; - mod[2] = (lpd_mode & 4) ? 1 : 0; - mod[3] = (lpd_mode & 8) ? 1 : 0; - break; - } - break; - } - } - return AAC_DEC_OK; -} - -static void CLpd_Reset( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - int keep_past_signal) { - int i; - - /* Reset TCX / ACELP common memory */ - if (!keep_past_signal) { - FDKmemclear(pAacDecoderStaticChannelInfo->old_synth, - sizeof(pAacDecoderStaticChannelInfo->old_synth)); - } - - /* Initialize the LSFs */ - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - pAacDecoderStaticChannelInfo->lpc4_lsf[i] = fdk_dec_lsf_init[i]; - } - - /* Reset memory needed by bass post-filter */ - FDKmemclear(pAacDecoderStaticChannelInfo->mem_bpf, - sizeof(pAacDecoderStaticChannelInfo->mem_bpf)); - - pAacDecoderStaticChannelInfo->old_bpf_control_info = 0; - for (i = 0; i < SYN_SFD; i++) { - pAacDecoderStaticChannelInfo->old_T_pf[i] = 64; - pAacDecoderStaticChannelInfo->old_gain_pf[i] = (FIXP_DBL)0; - } - - /* Reset ACELP memory */ - CLpd_AcelpReset(&pAacDecoderStaticChannelInfo->acelp); - - pAacDecoderStaticChannelInfo->last_lpc_lost = 0; /* prev_lpc_lost */ - pAacDecoderStaticChannelInfo->last_tcx_pitch = L_DIV; /* pitch_tcx */ - pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; /* nbLostCmpt */ -} - -/* - * Externally visible functions - */ - -AAC_DECODER_ERROR CLpdChannelStream_Read( - HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, UINT flags) { - AAC_DECODER_ERROR error = AAC_DEC_OK; - int first_tcx_flag; - int k, nbDiv, fFacDataPresent, first_lpd_flag, acelp_core_mode, - facGetMemState = 0; - UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; - int lpd_mode_last, prev_frame_was_lpd; - USAC_COREMODE core_mode_last; - const int lg_table_offset = 0; - const int *lg_table = (pAacDecoderChannelInfo->granuleLength == 128) - ? &lg_table_ccfl[0][lg_table_offset] - : &lg_table_ccfl[1][lg_table_offset]; - int last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; - - int last_frame_ok = 1; - - INT i_offset; - UINT samplingRate; - - samplingRate = pSamplingRateInfo->samplingRate; - - i_offset = - (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - - (INT)PIT_MIN_12k8; - - if ((samplingRate < FAC_FSCALE_MIN) || (samplingRate > FAC_FSCALE_MAX)) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - - acelp_core_mode = FDKreadBits(hBs, 3); - - /* lpd_mode */ - error = CLpd_ReadAndMapLpdModeToModArray(mod, hBs, 0); - if (error != AAC_DEC_OK) { - goto bail; - } - - /* bpf_control_info */ - pAacDecoderChannelInfo->data.usac.bpf_control_info = FDKreadBit(hBs); - - /* last_core_mode */ - prev_frame_was_lpd = FDKreadBit(hBs); - /* fac_data_present */ - fFacDataPresent = FDKreadBit(hBs); - - /* Set valid values from - * pAacDecoderStaticChannelInfo->{last_core_mode,last_lpd_mode} */ - pAacDecoderChannelInfo->data.usac.core_mode_last = - pAacDecoderStaticChannelInfo->last_core_mode; - lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last = - pAacDecoderStaticChannelInfo->last_lpd_mode; - - if (prev_frame_was_lpd == 0) { - /* Last frame was FD */ - pAacDecoderChannelInfo->data.usac.core_mode_last = FD_LONG; - pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; - } else { - /* Last frame was LPD */ - pAacDecoderChannelInfo->data.usac.core_mode_last = LPD; - if (((mod[0] == 0) && fFacDataPresent) || - ((mod[0] != 0) && !fFacDataPresent)) { - /* Currend mod is ACELP, fac data present -> TCX, current mod TCX, no fac - * data -> TCX */ - if (lpd_mode_last == 0) { - /* Bit stream interruption detected. Assume last TCX mode as TCX20. */ - pAacDecoderChannelInfo->data.usac.lpd_mode_last = 1; - } - /* Else assume that remembered TCX mode is correct. */ - } else { - pAacDecoderChannelInfo->data.usac.lpd_mode_last = 0; - } - } - - first_lpd_flag = (pAacDecoderChannelInfo->data.usac.core_mode_last != - LPD); /* Depends on bitstream configuration */ - first_tcx_flag = 1; - - if (pAacDecoderStaticChannelInfo->last_core_mode != - LPD) { /* ATTENTION: Reset depends on what we rendered before! */ - CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 0); - - if (!last_frame_ok) { - /* If last rendered frame was not LPD and first lpd flag is not set, this - * must be an error - set last_lpc_lost flag */ - last_lpc_lost |= (first_lpd_flag) ? 0 : 1; - } - } - - core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last; - lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last; - - nbDiv = NB_DIV; - - /* k is the frame index. If a frame is of size 40MS or 80MS, - this frame index is incremented 2 or 4 instead of 1 respectively. */ - - k = 0; - while (k < nbDiv) { - /* Reset FAC data pointers in order to avoid applying old random FAC data. - */ - pAacDecoderChannelInfo->data.usac.fac_data[k] = NULL; - - if ((k == 0 && core_mode_last == LPD && fFacDataPresent) || - (lpd_mode_last == 0 && mod[k] > 0) || - ((lpd_mode_last != 255) && lpd_mode_last > 0 && mod[k] == 0)) { - int err; - - /* Assign FAC memory */ - pAacDecoderChannelInfo->data.usac.fac_data[k] = - CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); - - /* FAC for (ACELP -> TCX) or (TCX -> ACELP) */ - err = CLpd_FAC_Read( - hBs, pAacDecoderChannelInfo->data.usac.fac_data[k], - pAacDecoderChannelInfo->data.usac.fac_data_e, - pAacDecoderChannelInfo->granuleLength, /* == fac_length */ - 0, k); - if (err != 0) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - } - - if (mod[k] == 0) /* acelp-mode */ - { - int err; - err = CLpd_AcelpRead( - hBs, &pAacDecoderChannelInfo->data.usac.acelp[k], acelp_core_mode, - pAacDecoderChannelInfo->granuleLength * 8 /* coreCoderFrameLength */, - i_offset); - if (err != 0) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - - lpd_mode_last = 0; - k++; - } else /* mode != 0 => TCX */ - { - error = CLpd_TCX_Read(hBs, pAacDecoderChannelInfo, - pAacDecoderStaticChannelInfo, lg_table[mod[k]], - first_tcx_flag, k, flags); - - lpd_mode_last = mod[k]; - first_tcx_flag = 0; - k += 1 << (mod[k] - 1); - } - if (error != AAC_DEC_OK) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - } - - { - int err; - - /* Read LPC coefficients */ - err = CLpc_Read( - hBs, pAacDecoderChannelInfo->data.usac.lsp_coeff, - pAacDecoderStaticChannelInfo->lpc4_lsf, - pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, - pAacDecoderChannelInfo->data.usac.aStability, mod, first_lpd_flag, - /* if last lpc4 is available from concealment do not extrapolate lpc0 - from lpc2 */ - (mod[0] & 0x3) ? 0 - : (last_lpc_lost && - pAacDecoderStaticChannelInfo->last_core_mode != LPD), - last_frame_ok); - if (err != 0) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - } - - /* adjust old lsp[] following to a bad frame (to avoid overshoot) (ref: - * dec_LPD.c) */ - if (last_lpc_lost && !last_frame_ok) { - int k_next; - k = 0; - while (k < nbDiv) { - int i; - k_next = k + (((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1))); - FIXP_LPC *lsp_old = pAacDecoderChannelInfo->data.usac.lsp_coeff[k]; - FIXP_LPC *lsp_new = pAacDecoderChannelInfo->data.usac.lsp_coeff[k_next]; - - for (i = 0; i < M_LP_FILTER_ORDER; i++) { - if (lsp_new[i] < lsp_old[i]) { - lsp_old[i] = lsp_new[i]; - } - } - k = k_next; - } - } - - if (pAacDecoderStaticChannelInfo->last_lpd_mode != 0) { - if (pAacDecoderStaticChannelInfo->last_lpd_mode == 255) { - /* We need it for TCX decoding or ACELP excitation update */ - E_LPC_f_lsp_a_conversion( - pAacDecoderChannelInfo->data.usac.lsp_coeff[0], - pAacDecoderChannelInfo->data.usac.lp_coeff[0], - &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); - } else { /* last_lpd_mode was TCX */ - /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid - * converting LSP coefficients again). */ - FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], - pAacDecoderStaticChannelInfo->lp_coeff_old[0], - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = - pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; - } - } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ - - if (fFacDataPresent && (core_mode_last != LPD)) { - int prev_frame_was_short; - - prev_frame_was_short = FDKreadBit(hBs); - - if (prev_frame_was_short) { - core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last = - FD_SHORT; - pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; - - if ((pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT)) { - /* USAC Conformance document: - short_fac_flag shall be encoded with a value of 1 if the - window_sequence of the previous frame was 2 (EIGHT_SHORT_SEQUENCE). - Otherwise short_fac_flag shall be encoded with a - value of 0. */ - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - } - - /* Assign memory */ - pAacDecoderChannelInfo->data.usac.fac_data[0] = - CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); - - { - int err; - - /* FAC for FD -> ACELP */ - err = CLpd_FAC_Read( - hBs, pAacDecoderChannelInfo->data.usac.fac_data[0], - pAacDecoderChannelInfo->data.usac.fac_data_e, - CLpd_FAC_getLength(core_mode_last != FD_SHORT, - pAacDecoderChannelInfo->granuleLength), - 1, 0); - if (err != 0) { - error = AAC_DEC_PARSE_ERROR; - goto bail; - } - } - } - -bail: - if (error == AAC_DEC_OK) { - /* check consitency of last core/lpd mode values */ - if ((pAacDecoderChannelInfo->data.usac.core_mode_last != - pAacDecoderStaticChannelInfo->last_core_mode) && - (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { - /* Something got wrong! */ - /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ - } else if ((pAacDecoderChannelInfo->data.usac.core_mode_last == LPD) && - (pAacDecoderChannelInfo->data.usac.lpd_mode_last != - pAacDecoderStaticChannelInfo->last_lpd_mode) && - (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { - /* Something got wrong! */ - /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ - } - } - - return error; -} - -void CLpdChannelStream_Decode( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags) { - UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; - int k; - UCHAR last_lpd_mode; - int nbDiv = NB_DIV; - - /* k is the frame index. If a frame is of size 40MS or 80MS, - this frame index is incremented 2 or 4 instead of 1 respectively. */ - k = 0; - last_lpd_mode = - pAacDecoderChannelInfo->data.usac - .lpd_mode_last; /* could be different to what has been rendered */ - while (k < nbDiv) { - if (mod[k] == 0) { - /* ACELP */ - - /* If FAC (fac_data[k] != NULL), and previous frame was TCX, apply (TCX) - * gains to FAC data */ - if (last_lpd_mode > 0 && last_lpd_mode != 255 && - pAacDecoderChannelInfo->data.usac.fac_data[k]) { - CFac_ApplyGains(pAacDecoderChannelInfo->data.usac.fac_data[k], - pAacDecoderChannelInfo->granuleLength, - pAacDecoderStaticChannelInfo->last_tcx_gain, - pAacDecoderStaticChannelInfo->last_alfd_gains, - (last_lpd_mode < 4) ? last_lpd_mode : 3); - - pAacDecoderChannelInfo->data.usac.fac_data_e[k] += - pAacDecoderStaticChannelInfo->last_tcx_gain_e; - } - } else { - /* TCX */ - CLpd_TcxDecode(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, - flags, mod[k], last_lpd_mode, k, 1 /* frameOk == 1 */ - ); - - /* Store TCX gain scale for next possible FAC transition. */ - pAacDecoderStaticChannelInfo->last_tcx_gain = - pAacDecoderChannelInfo->data.usac.tcx_gain[k]; - pAacDecoderStaticChannelInfo->last_tcx_gain_e = - pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; - - /* If FAC (fac_data[k] != NULL), apply gains */ - if (last_lpd_mode == 0 && pAacDecoderChannelInfo->data.usac.fac_data[k]) { - CFac_ApplyGains( - pAacDecoderChannelInfo->data.usac.fac_data[k], - pAacDecoderChannelInfo->granuleLength /* == fac_length */, - pAacDecoderChannelInfo->data.usac.tcx_gain[k], - pAacDecoderStaticChannelInfo->last_alfd_gains, mod[k]); - - pAacDecoderChannelInfo->data.usac.fac_data_e[k] += - pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; - } - } - - /* remember previous mode */ - last_lpd_mode = mod[k]; - - /* Increase k to next frame */ - k += (mod[k] == 0) ? 1 : (1 << (mod[k] - 1)); - } -} - -AAC_DECODER_ERROR CLpd_RenderTimeSignal( - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, - INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags, - UINT strmFlags) { - UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; - AAC_DECODER_ERROR error = AAC_DEC_OK; - int k, i_offset; - int last_k; - int nrSamples = 0; - int facFB = 1; - int nbDiv = NB_DIV; - int lDiv = lFrame / nbDiv; /* length of division (acelp or tcx20 frame)*/ - int lFac = lDiv / 2; - int nbSubfr = - lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ - int nbSubfrSuperfr = nbDiv * nbSubfr; - int synSfd = (nbSubfrSuperfr / 2) - BPF_SFD; - int SynDelay = synSfd * L_SUBFR; - int aacDelay = lFrame / 2; - - /* - In respect to the reference software, the synth pointer here is lagging by - aacDelay ( == SYN_DELAY + BPF_DELAY ) samples. The corresponding old - synthesis samples are handled by the IMDCT overlap. - */ - - FIXP_DBL *synth_buf = - pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->synth_buf; - FIXP_DBL *synth = synth_buf + PIT_MAX_MAX - BPF_DELAY; - UCHAR last_lpd_mode, last_last_lpd_mode, last_lpc_lost, last_frame_lost; - - INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; - FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; - - const int *lg_table; - int lg_table_offset = 0; - - UINT samplingRate = pSamplingRateInfo->samplingRate; - - FDKmemclear(pitch, (NB_SUBFR_SUPERFR + SYN_SFD) * sizeof(INT)); - - if (flags & AACDEC_FLUSH) { - CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, - flags & AACDEC_FLUSH); - frameOk = 0; - } - - switch (lFrame) { - case 1024: - lg_table = &lg_table_ccfl[0][lg_table_offset]; - break; - case 768: - lg_table = &lg_table_ccfl[1][lg_table_offset]; - break; - default: - FDK_ASSERT(0); - return AAC_DEC_UNKNOWN; - } - - last_frame_lost = 0; - - /* Maintain LPD mode from previous frame */ - if ((pAacDecoderStaticChannelInfo->last_core_mode == FD_LONG) || - (pAacDecoderStaticChannelInfo->last_core_mode == FD_SHORT)) { - pAacDecoderStaticChannelInfo->last_lpd_mode = 255; - } - - if (!frameOk) { - FIXP_DBL old_tcx_gain; - FIXP_SGL old_stab; - SCHAR old_tcx_gain_e; - int nLostSf; - - last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; - old_tcx_gain = pAacDecoderStaticChannelInfo->last_tcx_gain; - old_tcx_gain_e = pAacDecoderStaticChannelInfo->last_tcx_gain_e; - old_stab = pAacDecoderStaticChannelInfo->oldStability; - nLostSf = pAacDecoderStaticChannelInfo->numLostLpdFrames; - - /* patch the last LPD mode */ - pAacDecoderChannelInfo->data.usac.lpd_mode_last = last_lpd_mode; - - /* Do mode extrapolation and repeat the previous mode: - if previous mode = ACELP -> ACELP - if previous mode = TCX-20/40 -> TCX-20 - if previous mode = TCX-80 -> TCX-80 - notes: - - ACELP is not allowed after TCX (no pitch information to reuse) - - TCX-40 is not allowed in the mode repetition to keep the logic simple - */ - switch (last_lpd_mode) { - case 0: - mod[0] = mod[1] = mod[2] = mod[3] = 0; /* -> ACELP concealment */ - break; - case 3: - mod[0] = mod[1] = mod[2] = mod[3] = 3; /* -> TCX FD concealment */ - break; - case 2: - mod[0] = mod[1] = mod[2] = mod[3] = 2; /* -> TCX FD concealment */ - break; - case 1: - default: - mod[0] = mod[1] = mod[2] = mod[3] = 4; /* -> TCX TD concealment */ - break; - } - - /* LPC extrapolation */ - CLpc_Conceal(pAacDecoderChannelInfo->data.usac.lsp_coeff, - pAacDecoderStaticChannelInfo->lpc4_lsf, - pAacDecoderStaticChannelInfo->lsf_adaptive_mean, - /*(pAacDecoderStaticChannelInfo->numLostLpdFrames == 0) ||*/ - (last_lpd_mode == 255)); - - if ((last_lpd_mode > 0) && (last_lpd_mode < 255)) { - /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid - * converting LSP coefficients again). */ - FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], - pAacDecoderStaticChannelInfo->lp_coeff_old[0], - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = - pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; - } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ - /* case last_lpd_mode was Time domain TCX concealment is handled after this - * "if (!frameOk)"-block */ - - /* k is the frame index. If a frame is of size 40MS or 80MS, - this frame index is incremented 2 or 4 instead of 1 respectively. */ - k = 0; - while (k < nbDiv) { - pAacDecoderChannelInfo->data.usac.tcx_gain[k] = old_tcx_gain; - pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] = old_tcx_gain_e; - - /* restore stability value from last frame */ - pAacDecoderChannelInfo->data.usac.aStability[k] = old_stab; - - /* Increase k to next frame */ - k += ((mod[k] & 0x3) == 0) ? 1 : (1 << ((mod[k] & 0x3) - 1)); - - nLostSf++; - } - } else { - if ((pAacDecoderStaticChannelInfo->last_lpd_mode == 4) && (mod[0] > 0)) { - /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid - * converting LSP coefficients again). */ - FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], - pAacDecoderStaticChannelInfo->lp_coeff_old[0], - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = - pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; - } - } - - Acelp_PreProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, pitch, - pAacDecoderStaticChannelInfo->old_T_pf, pit_gain, - pAacDecoderStaticChannelInfo->old_gain_pf, samplingRate, - &i_offset, lFrame, synSfd, nbSubfrSuperfr); - - /* k is the frame index. If a frame is of size 40MS or 80MS, - this frame index is incremented 2 or 4 instead of 1 respectively. */ - k = 0; - last_k = -1; /* mark invalid */ - last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; - last_last_lpd_mode = pAacDecoderStaticChannelInfo->last_last_lpd_mode; - last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost | last_frame_lost; - - /* This buffer must be avalable for the case of FD->ACELP transition. The - beginning of the buffer is used after the BPF to overwrite the output signal. - Only the FAC area must be affected by the BPF */ - - while (k < nbDiv) { - if (frameOk == 0) { - pAacDecoderStaticChannelInfo->numLostLpdFrames++; - } else { - last_frame_lost |= - (pAacDecoderStaticChannelInfo->numLostLpdFrames > 0) ? 1 : 0; - pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; - } - if (mod[k] == 0 || mod[k] == 4) { - /* ACELP or TCX time domain concealment */ - FIXP_DBL *acelp_out; - - /* FAC management */ - if ((last_lpd_mode != 0) && (last_lpd_mode != 4)) /* TCX TD concealment */ - { - FIXP_DBL *pFacData = NULL; - - if (frameOk && !last_frame_lost) { - pFacData = pAacDecoderChannelInfo->data.usac.fac_data[k]; - } - - nrSamples += CLpd_FAC_Mdct2Acelp( - &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, pFacData, - pAacDecoderChannelInfo->data.usac.fac_data_e[k], - pAacDecoderChannelInfo->data.usac.lp_coeff[k], - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], - lFrame - nrSamples, - CLpd_FAC_getLength( - (pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) || - (k > 0), - lFac), - (pAacDecoderStaticChannelInfo->last_core_mode != LPD) && (k == 0), - 0); - - FDKmemcpy( - synth + nrSamples, pAacDecoderStaticChannelInfo->IMdct.overlap.time, - pAacDecoderStaticChannelInfo->IMdct.ov_offset * sizeof(FIXP_DBL)); - { - FIXP_LPC *lp_prev = - pAacDecoderChannelInfo->data.usac - .lp_coeff[0]; /* init value does not real matter */ - INT lp_prev_exp = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]; - - if (last_lpd_mode != 255) { /* last mode was tcx */ - last_k = k - (1 << (last_lpd_mode - 1)); - if (last_k < 0) { - lp_prev = pAacDecoderStaticChannelInfo->lp_coeff_old[1]; - lp_prev_exp = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1]; - } else { - lp_prev = pAacDecoderChannelInfo->data.usac.lp_coeff[last_k]; - lp_prev_exp = - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; - } - } - - CLpd_AcelpPrepareInternalMem( - synth + aacDelay + k * lDiv, last_lpd_mode, - (last_last_lpd_mode == 4) ? 0 : last_last_lpd_mode, - pAacDecoderChannelInfo->data.usac.lp_coeff[k], - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], lp_prev, - lp_prev_exp, &pAacDecoderStaticChannelInfo->acelp, lFrame, - (last_frame_lost && k < 2), mod[k]); - } - } else { - if (k == 0 && pAacDecoderStaticChannelInfo->IMdct.ov_offset != - lFrame / facFB / 2) { - pAacDecoderStaticChannelInfo->IMdct.ov_offset = lFrame / facFB / 2; - } - nrSamples += imdct_drain(&pAacDecoderStaticChannelInfo->IMdct, - synth + nrSamples, lFrame / facFB - nrSamples); - } - - if (nrSamples >= lFrame / facFB) { - /* Write ACELP time domain samples into IMDCT overlap buffer at - * pAacDecoderStaticChannelInfo->IMdct.overlap.time + - * pAacDecoderStaticChannelInfo->IMdct.ov_offset - */ - acelp_out = pAacDecoderStaticChannelInfo->IMdct.overlap.time + - pAacDecoderStaticChannelInfo->IMdct.ov_offset; - - /* Account ACELP time domain output samples to overlap buffer */ - pAacDecoderStaticChannelInfo->IMdct.ov_offset += lDiv; - } else { - /* Write ACELP time domain samples into output buffer at pTimeData + - * nrSamples */ - acelp_out = synth + nrSamples; - - /* Account ACELP time domain output samples to output buffer */ - nrSamples += lDiv; - } - - if (mod[k] == 4) { - pAacDecoderStaticChannelInfo->acelp.wsyn_rms = scaleValue( - pAacDecoderChannelInfo->data.usac.tcx_gain[k], - fixMin(0, - pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] - SF_EXC)); - CLpd_TcxTDConceal(&pAacDecoderStaticChannelInfo->acelp, - &pAacDecoderStaticChannelInfo->last_tcx_pitch, - pAacDecoderChannelInfo->data.usac.lsp_coeff[k], - pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], - pAacDecoderChannelInfo->data.usac.aStability[k], - pAacDecoderStaticChannelInfo->numLostLpdFrames, - acelp_out, lFrame, - pAacDecoderStaticChannelInfo->last_tcx_noise_factor); - - } else { - FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[k] >= - (FIXP_SGL)0); - CLpd_AcelpDecode(&pAacDecoderStaticChannelInfo->acelp, i_offset, - pAacDecoderChannelInfo->data.usac.lsp_coeff[k], - pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], - pAacDecoderChannelInfo->data.usac.aStability[k], - &pAacDecoderChannelInfo->data.usac.acelp[k], - pAacDecoderStaticChannelInfo->numLostLpdFrames, - last_lpc_lost, k, acelp_out, - &pitch[(k * nbSubfr) + synSfd], - &pit_gain[(k * nbSubfr) + synSfd], lFrame); - } - - if (mod[k] != 4) { - if (last_lpd_mode != 0 && - pAacDecoderChannelInfo->data.usac - .bpf_control_info) { /* FD/TCX -> ACELP transition */ - /* bass post-filter past FAC area (past two (one for FD short) - * subframes) */ - int currentSf = synSfd + k * nbSubfr; - - if ((k > 0) || (pAacDecoderStaticChannelInfo->last_core_mode != - FD_SHORT)) { /* TCX or FD long -> ACELP */ - pitch[currentSf - 2] = pitch[currentSf - 1] = pitch[currentSf]; - pit_gain[currentSf - 2] = pit_gain[currentSf - 1] = - pit_gain[currentSf]; - } else { /* FD short -> ACELP */ - pitch[currentSf - 1] = pitch[currentSf]; - pit_gain[currentSf - 1] = pit_gain[currentSf]; - } - } - } - } else { /* TCX */ - int lg = lg_table[mod[k]]; - int isFullBandLpd = 0; - - /* FAC management */ - if ((last_lpd_mode == 0) || (last_lpd_mode == 4)) /* TCX TD concealment */ - { - C_AALLOC_SCRATCH_START(fac_buf, FIXP_DBL, 1024 / 8); - - /* pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL means no FAC - * data available. */ - if (last_frame_lost == 1 || - pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL) { - FDKmemclear(fac_buf, 1024 / 8 * sizeof(FIXP_DBL)); - pAacDecoderChannelInfo->data.usac.fac_data[k] = fac_buf; - pAacDecoderChannelInfo->data.usac.fac_data_e[k] = 0; - } - - nrSamples += CLpd_FAC_Acelp2Mdct( - &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, - SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, - pAacDecoderChannelInfo->granuleLength, isFullBandLpd), - pAacDecoderChannelInfo->specScale + k, 1, - pAacDecoderChannelInfo->data.usac.fac_data[k], - pAacDecoderChannelInfo->data.usac.fac_data_e[k], - pAacDecoderChannelInfo->granuleLength /* == fac_length */, - lFrame - nrSamples, lg, - FDKgetWindowSlope(lDiv, - GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - lDiv, pAacDecoderChannelInfo->data.usac.lp_coeff[k], - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], - &pAacDecoderStaticChannelInfo->acelp, - pAacDecoderChannelInfo->data.usac.tcx_gain[k], - (last_frame_lost || !frameOk), 0 /* is not FD FAC */ - , - last_lpd_mode, k, - pAacDecoderChannelInfo - ->currAliasingSymmetry /* Note: The current aliasing - symmetry for a TCX (i.e. LPD) - frame must always be 0 */ - ); - - pitch[(k * nbSubfr) + synSfd + 1] = pitch[(k * nbSubfr) + synSfd] = - pitch[(k * nbSubfr) + synSfd - 1]; - pit_gain[(k * nbSubfr) + synSfd + 1] = - pit_gain[(k * nbSubfr) + synSfd] = - pit_gain[(k * nbSubfr) + synSfd - 1]; - - C_AALLOC_SCRATCH_END(fac_buf, FIXP_DBL, 1024 / 8); - } else { - int tl = lg; - int fl = lDiv; - int fr = lDiv; - - nrSamples += imlt_block( - &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, - SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, - pAacDecoderChannelInfo->granuleLength, isFullBandLpd), - pAacDecoderChannelInfo->specScale + k, 1, lFrame - nrSamples, tl, - FDKgetWindowSlope(fl, - GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - fl, - FDKgetWindowSlope(fr, - GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), - fr, pAacDecoderChannelInfo->data.usac.tcx_gain[k], - pAacDecoderChannelInfo->currAliasingSymmetry - ? MLT_FLAG_CURR_ALIAS_SYMMETRY - : 0); - } - } - /* remember previous mode */ - last_last_lpd_mode = last_lpd_mode; - last_lpd_mode = mod[k]; - last_lpc_lost = (frameOk == 0) ? 1 : 0; - - /* Increase k to next frame */ - last_k = k; - k += ((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1)); - } - - if (frameOk) { - /* assume data was ok => store for concealment */ - FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[last_k] >= - (FIXP_SGL)0); - pAacDecoderStaticChannelInfo->oldStability = - pAacDecoderChannelInfo->data.usac.aStability[last_k]; - FDKmemcpy(pAacDecoderStaticChannelInfo->lsf_adaptive_mean, - pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - } - - /* store past lp coeffs for next superframe (they are only valid and needed if - * last_lpd_mode was tcx) */ - if (last_lpd_mode > 0) { - FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[0], - pAacDecoderChannelInfo->data.usac.lp_coeff[nbDiv], - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0] = - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[nbDiv]; - FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[1], - pAacDecoderChannelInfo->data.usac.lp_coeff[last_k], - M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); - pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1] = - pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; - } - - FDK_ASSERT(nrSamples == lFrame); - - /* check whether usage of bass postfilter was de-activated in the bitstream; - if yes, set pitch gain to 0 */ - if (!(pAacDecoderChannelInfo->data.usac.bpf_control_info)) { - if (mod[0] != 0 && (pAacDecoderStaticChannelInfo->old_bpf_control_info)) { - for (int i = 2; i < nbSubfrSuperfr; i++) - pit_gain[synSfd + i] = (FIXP_DBL)0; - } else { - for (int i = 0; i < nbSubfrSuperfr; i++) - pit_gain[synSfd + i] = (FIXP_DBL)0; - } - } - - /* for bass postfilter */ - for (int n = 0; n < synSfd; n++) { - pAacDecoderStaticChannelInfo->old_T_pf[n] = pitch[nbSubfrSuperfr + n]; - pAacDecoderStaticChannelInfo->old_gain_pf[n] = pit_gain[nbSubfrSuperfr + n]; - } - - pAacDecoderStaticChannelInfo->old_bpf_control_info = - pAacDecoderChannelInfo->data.usac.bpf_control_info; - - { - INT lookahead = -BPF_DELAY; - int copySamp = (mod[nbDiv - 1] == 0) ? (aacDelay) : (aacDelay - lFac); - - /* Copy enough time domain samples from MDCT to synthesis buffer as needed - * by the bass postfilter */ - - lookahead += imdct_copy_ov_and_nr(&pAacDecoderStaticChannelInfo->IMdct, - synth + nrSamples, copySamp); - - FDK_ASSERT(lookahead == copySamp - BPF_DELAY); - - FIXP_DBL *p2_synth = synth + BPF_DELAY; - - /* recalculate pitch gain to allow postfilering on FAC area */ - for (int i = 0; i < nbSubfrSuperfr; i++) { - int T = pitch[i]; - FIXP_DBL gain = pit_gain[i]; - - if (gain > (FIXP_DBL)0) { - gain = get_gain(&p2_synth[i * L_SUBFR], &p2_synth[(i * L_SUBFR) - T], - L_SUBFR); - pit_gain[i] = gain; - } - } - - { - bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB, - mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay, - pTimeData, pAacDecoderStaticChannelInfo->mem_bpf); - } - } - - Acelp_PostProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, - pitch, pAacDecoderStaticChannelInfo->old_T_pf, lFrame, - synSfd, nbSubfrSuperfr); - - /* Store last mode for next super frame */ - { pAacDecoderStaticChannelInfo->last_core_mode = LPD; } - pAacDecoderStaticChannelInfo->last_lpd_mode = last_lpd_mode; - pAacDecoderStaticChannelInfo->last_last_lpd_mode = last_last_lpd_mode; - pAacDecoderStaticChannelInfo->last_lpc_lost = last_lpc_lost; - - return error; -} diff --git a/libAACdec/src/usacdec_lpd.h b/libAACdec/src/usacdec_lpd.h deleted file mode 100644 index 3e7938d..0000000 --- a/libAACdec/src/usacdec_lpd.h +++ /dev/null @@ -1,198 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): Manuel Jander - - Description: USAC Linear Prediction Domain coding - -*******************************************************************************/ - -#ifndef USACDEC_LPD_H -#define USACDEC_LPD_H - -#include "channelinfo.h" - -#define OPTIMIZE_AVG_PERFORMANCE - -/** - * \brief read a lpd_channel_stream. - * \param hBs a bit stream handle, where the lpd_channel_stream is located. - * \param pAacDecoderChannelInfo the channel context structure for storing read - * data. - * \param flags bit stream syntax flags. - * \return AAC_DECODER_ERROR error code. - */ -AAC_DECODER_ERROR CLpdChannelStream_Read( - HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - const SamplingRateInfo *pSamplingRateInfo, UINT flags); - -/** - * \brief decode one lpd_channel_stream and render the audio output. - * \param pAacDecoderChannelInfo struct holding the channel information to be - * rendered. - * \param pAacDecoderStaticChannelInfo struct holding the persistent channel - * information to be rendered. - * \param pSamplingRateInfo holds the sampling rate information - * \param elFlags holds the internal decoder flags - */ -void CLpdChannelStream_Decode( - CAacDecoderChannelInfo *pAacDecoderChannelInfo, - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags); - -/** - * \brief generate time domain output signal for LPD channel streams - * \param pAacDecoderStaticChannelInfo - * \param pAacDecoderChannelInfo - * \param pTimeData pointer to output buffer - * \param samplesPerFrame amount of output samples - * \param pSamplingRateInfo holds the sampling rate information - * \param pWorkBuffer1 pointer to work buffer for temporal data - */ -AAC_DECODER_ERROR CLpd_RenderTimeSignal( - CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, - CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, - INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, - UINT flags, UINT strmFlags); - -static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) { - if (fNotShortBlock) { - return (fac_length_long); - } else { - return fac_length_long / 2; - } -} - -void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, - const FIXP_SGL *filt, INT stop, int len); - -/** - * \brief perform a low-frequency pitch enhancement on time domain signal - * \param[in] syn pointer to time domain input signal - * \param[in] synFB pointer to time domain input signal - * \param[in] upsampling factor - * \param[in] T_sf array with past decoded pitch period values for each subframe - * \param[in] non_zero_gain_flags indicates whether pitch gains of past - * subframes are zero or not, msb -> [1 BPF_DELAY subfr][7 SYN_DELAY subfr][16 - * new subfr] <- lsb - * \param[in] l_frame length of filtering, must be multiple of L_SUBFR - * \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next - * \param[out] synth_out pointer to time domain output signal - * \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR) - */ - -void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain, - const int frame_length, const INT l_frame, - const INT l_next, FIXP_PCM *synth_out, - FIXP_DBL mem_bpf[]); - -/** - * \brief random sign generator for FD and TCX noise filling - * \param[in,out] seed pointer to random seed - * \return if return value is zero use positive sign - * \Note: This code is also implemented as a copy in block.cpp, grep for - * "UsacRandomSign" - */ -FDK_INLINE -int UsacRandomSign(ULONG *seed) { - *seed = (ULONG)((UINT64)(*seed) * 69069 + 5); - - return (int)((*seed) & 0x10000); -} - -void CFdp_Reset(CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo); - -#endif /* USACDEC_LPD_H */ diff --git a/libAACdec/src/usacdec_rom.cpp b/libAACdec/src/usacdec_rom.cpp deleted file mode 100644 index ca3009e..0000000 --- a/libAACdec/src/usacdec_rom.cpp +++ /dev/null @@ -1,1504 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): M. Jander - - Description: - -*******************************************************************************/ - -#include "usacdec_rom.h" - -#define NB_SPHERE 32 -#define NB_LEADER 37 -#define NB_LDSIGN 226 -#define NB_LDQ3 9 -#define NB_LDQ4 28 - -/* For bass post filter */ -#define FL2FXCONST_SGL_FILT(a) FL2FXCONST_SGL(a*(1 << SF_FILT_LP)) -#define SF_FILT_LP 1 - -/* table of factorial */ -const UINT fdk_dec_tab_factorial[8] = {5040, 720, 120, 24, 6, 2, 1, 1}; - -/* Da - Absolute leaders */ -const UCHAR fdk_dec_Da[NB_LEADER][8] = { - {1, 1, 1, 1, 1, 1, 1, 1}, {2, 2, 0, 0, 0, 0, 0, 0}, - {2, 2, 2, 2, 0, 0, 0, 0}, {3, 1, 1, 1, 1, 1, 1, 1}, - {4, 0, 0, 0, 0, 0, 0, 0}, {2, 2, 2, 2, 2, 2, 0, 0}, - {3, 3, 1, 1, 1, 1, 1, 1}, {4, 2, 2, 0, 0, 0, 0, 0}, - {2, 2, 2, 2, 2, 2, 2, 2}, {3, 3, 3, 1, 1, 1, 1, 1}, - {4, 2, 2, 2, 2, 0, 0, 0}, {4, 4, 0, 0, 0, 0, 0, 0}, - {5, 1, 1, 1, 1, 1, 1, 1}, {3, 3, 3, 3, 1, 1, 1, 1}, - {4, 2, 2, 2, 2, 2, 2, 0}, {4, 4, 2, 2, 0, 0, 0, 0}, - {5, 3, 1, 1, 1, 1, 1, 1}, {6, 2, 0, 0, 0, 0, 0, 0}, - {4, 4, 4, 0, 0, 0, 0, 0}, {6, 2, 2, 2, 0, 0, 0, 0}, - {6, 4, 2, 0, 0, 0, 0, 0}, {7, 1, 1, 1, 1, 1, 1, 1}, - {8, 0, 0, 0, 0, 0, 0, 0}, {6, 6, 0, 0, 0, 0, 0, 0}, - {8, 2, 2, 0, 0, 0, 0, 0}, {8, 4, 0, 0, 0, 0, 0, 0}, - {9, 1, 1, 1, 1, 1, 1, 1}, {10, 2, 0, 0, 0, 0, 0, 0}, - {8, 8, 0, 0, 0, 0, 0, 0}, {10, 6, 0, 0, 0, 0, 0, 0}, - {12, 0, 0, 0, 0, 0, 0, 0}, {12, 4, 0, 0, 0, 0, 0, 0}, - {10, 10, 0, 0, 0, 0, 0, 0}, {14, 2, 0, 0, 0, 0, 0, 0}, - {12, 8, 0, 0, 0, 0, 0, 0}, {16, 0, 0, 0, 0, 0, 0, 0}, - {20, 0, 0, 0, 0, 0, 0, 0}}; - -/* Ds - Sign codes of all signed leaders */ -const UCHAR fdk_dec_Ds[NB_LDSIGN] = { - 0, 3, 15, 63, 255, 0, 64, 192, 0, 16, 48, 112, 240, 1, 7, - 31, 127, 128, 131, 143, 191, 0, 128, 0, 4, 12, 28, 60, 124, 252, - 0, 3, 15, 63, 65, 71, 95, 192, 195, 207, 255, 0, 32, 96, 128, - 160, 224, 0, 1, 3, 7, 15, 31, 63, 127, 255, 1, 7, 31, 32, - 35, 47, 97, 103, 127, 224, 227, 239, 0, 8, 24, 56, 120, 128, 136, - 152, 184, 248, 0, 64, 192, 0, 3, 15, 63, 129, 135, 159, 255, 0, - 3, 15, 17, 23, 48, 51, 63, 113, 119, 240, 243, 255, 0, 2, 6, - 14, 30, 62, 126, 128, 130, 134, 142, 158, 190, 254, 0, 16, 48, 64, - 80, 112, 192, 208, 240, 1, 7, 31, 64, 67, 79, 127, 128, 131, 143, - 191, 193, 199, 223, 0, 64, 128, 192, 0, 32, 96, 224, 0, 16, 48, - 112, 128, 144, 176, 240, 0, 32, 64, 96, 128, 160, 192, 224, 1, 7, - 31, 127, 128, 131, 143, 191, 0, 128, 0, 64, 192, 0, 32, 96, 128, - 160, 224, 0, 64, 128, 192, 0, 3, 15, 63, 129, 135, 159, 255, 0, - 64, 128, 192, 0, 64, 192, 0, 64, 128, 192, 0, 128, 0, 64, 128, - 192, 0, 64, 192, 0, 64, 128, 192, 0, 64, 128, 192, 0, 128, 0, - 128}; - -/* Ns - Number of signed leader associated to a given absolute leader */ -const UCHAR fdk_dec_Ns[NB_LEADER] = { - 5, 3, 5, 8, 2, 7, 11, 6, 9, 12, 10, 3, 8, 13, 14, 9, 14, 4, 4, - 8, 8, 8, 2, 3, 6, 4, 8, 4, 3, 4, 2, 4, 3, 4, 4, 2, 2}; - -/* Ia - Position of the first signed leader associated to an absolute leader */ -const UCHAR fdk_dec_Ia[NB_LEADER] = { - 0, 5, 8, 13, 21, 23, 30, 41, 47, 56, 68, 78, 81, - 89, 102, 116, 125, 139, 143, 147, 155, 163, 171, 173, 176, 182, - 186, 194, 198, 201, 205, 207, 211, 214, 218, 222, 224}; - -/* Is - Cardinalite offset of signed leaders */ -const USHORT fdk_dec_Is[NB_LDSIGN] = { - 0, 1, 29, 99, 127, 128, 156, 212, 256, 326, 606, - 1026, 1306, 1376, 1432, 1712, 1880, 1888, 1896, 2064, 2344, 240, - 248, 0, 28, 196, 616, 1176, 1596, 1764, 1792, 1820, 2240, - 2660, 2688, 3024, 4144, 4480, 4508, 4928, 5348, 2400, 2568, 2904, - 3072, 3240, 3576, 5376, 5377, 5385, 5413, 5469, 5539, 5595, 5623, - 5631, 5632, 5912, 6472, 6528, 6696, 8376, 9216, 10056, 11736, 11904, - 11960, 12520, 12800, 13080, 14200, 15880, 17000, 17280, 17560, 18680, 20360, - 21480, 3744, 3772, 3828, 21760, 21768, 21936, 22216, 22272, 22328, 22608, - 22776, 22784, 22854, 23274, 23344, 24464, 25584, 26004, 28524, 28944, 30064, - 31184, 31254, 31674, 31744, 31800, 32136, 32976, 34096, 34936, 35272, 35328, - 35384, 35720, 36560, 37680, 38520, 38856, 38912, 39332, 40172, 40592, 41432, - 43112, 43952, 44372, 45212, 45632, 45968, 47088, 47424, 47480, 48320, 49160, - 49216, 49272, 50112, 50952, 51008, 51344, 52464, 3856, 3912, 3968, 4024, - 52800, 52856, 53024, 53192, 53248, 53528, 54368, 55208, 55488, 55768, 56608, - 57448, 57728, 58064, 58400, 58736, 59072, 59408, 59744, 60080, 60416, 60472, - 60752, 60920, 60928, 60936, 61104, 61384, 4080, 4088, 61440, 61468, 61524, - 61552, 61720, 62056, 62224, 62392, 62728, 62896, 62952, 63008, 63064, 63120, - 63128, 63296, 63576, 63632, 63688, 63968, 64136, 64144, 64200, 64256, 64312, - 64368, 64396, 64452, 64480, 64536, 64592, 64648, 64704, 64712, 64720, 64776, - 64832, 64888, 64944, 64972, 65028, 65056, 65112, 65168, 65224, 65280, 65336, - 65392, 65448, 65504, 65512, 65520, 65528}; - -/* A3 - Number of the absolute leaders in codebooks Q2 and Q3 */ -const UCHAR fdk_dec_A3[NB_LDQ3] = {0, 1, 4, 2, 3, 7, 11, 17, 22}; - -/* A4 - Number of the absolute leaders in codebook Q4 */ -const UCHAR fdk_dec_A4[NB_LDQ4] = {5, 6, 8, 9, 10, 12, 13, 14, 15, 16, - 18, 19, 20, 21, 23, 24, 25, 26, 27, 28, - 29, 30, 31, 32, 33, 34, 35, 36}; - -/* I3 - Cardinality offsets for absolute leaders in Q3 */ -const USHORT fdk_dec_I3[NB_LDQ3] = {0, 128, 240, 256, 1376, - 2400, 3744, 3856, 4080}; - -/* I4 - Cardinality offset for absolute leaders in Q4 */ -const USHORT fdk_dec_I4[NB_LDQ4] = { - 0, 1792, 5376, 5632, 12800, 21760, 22784, 31744, 38912, 45632, - 52800, 53248, 57728, 60416, 61440, 61552, 62896, 63120, 64144, 64368, - 64480, 64704, 64720, 64944, 65056, 65280, 65504, 65520}; - -/* Initial ISF memory for concealment case */ -#define LSFI(x) ((x) << (FRACT_BITS - LSF_SCALE - 1)) - -const FIXP_LPC fdk_dec_lsf_init[16] = {1506, 3012, 4518, 6024, 7529, 9035, - 10541, 12047, 13553, 15059, 16565, 18071, - 19576, 21082, 22588, 24094}; - -/* dico_lsf_abs_8b is scaled by 1/(1<<13) */ -#define DICO(x) FX_DBL2FXCONST_LPC(x >> (LSF_SCALE - 13)) - -const FIXP_LPC fdk_dec_dico_lsf_abs_8b[] = { - DICO(0x05e57fe8), DICO(0x0ac00810), DICO(0x11ed8500), DICO(0x16d42ce0), - DICO(0x1beb1e20), DICO(0x217eaf40), DICO(0x2768c740), DICO(0x2d26f600), - DICO(0x32fe68c0), DICO(0x38b1d980), DICO(0x3e95bd80), DICO(0x446dab00), - DICO(0x4abfd280), DICO(0x5094b380), DICO(0x56ccb800), DICO(0x5c9aba00), - DICO(0x09660ca0), DICO(0x10ab4c00), DICO(0x15a16f20), DICO(0x19d3c780), - DICO(0x1ee99060), DICO(0x241d1200), DICO(0x29c83700), DICO(0x2f098f00), - DICO(0x34803fc0), DICO(0x3a37bc00), DICO(0x3ff55580), DICO(0x45da9280), - DICO(0x4bec6700), DICO(0x5169e300), DICO(0x57797c80), DICO(0x5d09ae80), - DICO(0x08a203b0), DICO(0x0d6ed1a0), DICO(0x152ccf20), DICO(0x19639dc0), - DICO(0x1d7e3e60), DICO(0x21f4a7c0), DICO(0x27b2f8c0), DICO(0x2dbb4480), - DICO(0x33ecde80), DICO(0x3982e100), DICO(0x3ea16100), DICO(0x43ab6080), - DICO(0x49534a80), DICO(0x4ea7e100), DICO(0x550d6300), DICO(0x5bcdcc80), - DICO(0x072dd048), DICO(0x0c654690), DICO(0x1436e940), DICO(0x19459680), - DICO(0x1e0041c0), DICO(0x2240dc80), DICO(0x26de4040), DICO(0x2b509b00), - DICO(0x309d8780), DICO(0x36151180), DICO(0x3c6c1200), DICO(0x42df6b80), - DICO(0x4a144400), DICO(0x50541280), DICO(0x56c34b80), DICO(0x5cb6c600), - DICO(0x051fef00), DICO(0x06b9fb48), DICO(0x0b4f9cc0), DICO(0x17e27800), - DICO(0x1b8c7340), DICO(0x1f772ca0), DICO(0x2478dc80), DICO(0x28242240), - DICO(0x2f27c640), DICO(0x33b03e80), DICO(0x381f20c0), DICO(0x3c662c00), - DICO(0x49565080), DICO(0x529b0f00), DICO(0x583ed080), DICO(0x5d8cec00), - DICO(0x071c4d18), DICO(0x097853b0), DICO(0x0f0f0690), DICO(0x157bf980), - DICO(0x1801f580), DICO(0x1deb0c20), DICO(0x2523da40), DICO(0x28534600), - DICO(0x2eb499c0), DICO(0x32eb5ac0), DICO(0x36749580), DICO(0x3a748200), - DICO(0x4325f700), DICO(0x515d8300), DICO(0x58a18700), DICO(0x5d722100), - DICO(0x06cbcd88), DICO(0x08bb6740), DICO(0x0dead310), DICO(0x152f0cc0), - DICO(0x18427640), DICO(0x1d9f2f20), DICO(0x22ba3b40), DICO(0x271a6e80), - DICO(0x2c677ec0), DICO(0x31061b00), DICO(0x349eef40), DICO(0x3c531b80), - DICO(0x4aed0580), DICO(0x4f8bbf80), DICO(0x54b74980), DICO(0x5bc9b700), - DICO(0x046410c8), DICO(0x06522ab0), DICO(0x0b6528c0), DICO(0x0f94bd90), - DICO(0x1a8f8b80), DICO(0x1ea57820), DICO(0x233ee180), DICO(0x27b3acc0), - DICO(0x2bd1d240), DICO(0x2fc4bcc0), DICO(0x3a98ea40), DICO(0x43d3f500), - DICO(0x49b37580), DICO(0x4e2afd00), DICO(0x55953300), DICO(0x5d36f600), - DICO(0x05d0f6c8), DICO(0x07e56d90), DICO(0x0be98080), DICO(0x0f956f30), - DICO(0x1259b3c0), DICO(0x1f08b240), DICO(0x25008c00), DICO(0x2900b180), - DICO(0x31ea6f00), DICO(0x352d1e00), DICO(0x3c970c80), DICO(0x45271200), - DICO(0x4b632280), DICO(0x5098a480), DICO(0x5672fc80), DICO(0x5c163180), - DICO(0x05bd81a0), DICO(0x07d4b8f0), DICO(0x0ce224b0), DICO(0x110abe20), - DICO(0x13dfeac0), DICO(0x17dedae0), DICO(0x2535c0c0), DICO(0x2a19da80), - DICO(0x2e5224c0), DICO(0x38ddeec0), DICO(0x3da99d80), DICO(0x42799100), - DICO(0x48973b00), DICO(0x4ea62880), DICO(0x53f77e80), DICO(0x5bd9c100), - DICO(0x0395cd50), DICO(0x058244b8), DICO(0x0af45520), DICO(0x1329cea0), - DICO(0x1a3970c0), DICO(0x1d9f2e00), DICO(0x21704400), DICO(0x277a34c0), - DICO(0x30215b40), DICO(0x33875040), DICO(0x3c159840), DICO(0x452fea00), - DICO(0x4981d200), DICO(0x4e15a980), DICO(0x54e84780), DICO(0x5c79ea00), - DICO(0x05413b98), DICO(0x08132a80), DICO(0x0dc7f050), DICO(0x13e25460), - DICO(0x1784bf80), DICO(0x1d630200), DICO(0x238bc880), DICO(0x28cc0880), - DICO(0x30da1a40), DICO(0x391e2200), DICO(0x415d8d00), DICO(0x48f13280), - DICO(0x4e300300), DICO(0x52e56580), DICO(0x5849fe80), DICO(0x5cdef400), - DICO(0x04a058c8), DICO(0x07569b88), DICO(0x0ef26610), DICO(0x13208140), - DICO(0x168c0500), DICO(0x1afec080), DICO(0x22a0abc0), DICO(0x2a057880), - DICO(0x2fd1c840), DICO(0x3703c680), DICO(0x3d326b80), DICO(0x43df2e80), - DICO(0x4a6f9000), DICO(0x50900d80), DICO(0x56c73f00), DICO(0x5cc3da80), - DICO(0x065c99e8), DICO(0x09060c50), DICO(0x0d1ef1c0), DICO(0x16bd9020), - DICO(0x1a04dae0), DICO(0x1e3c0580), DICO(0x25783700), DICO(0x29710ac0), - DICO(0x309cbb80), DICO(0x36c66280), DICO(0x3adb0580), DICO(0x41b37e00), - DICO(0x496ca700), DICO(0x4dab7600), DICO(0x52be6280), DICO(0x58fec480), - DICO(0x04640880), DICO(0x05a75ab8), DICO(0x0edba410), DICO(0x16e076a0), - DICO(0x198acec0), DICO(0x1eb5fae0), DICO(0x228c9000), DICO(0x29986c00), - 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DICO(0x2d374c00), DICO(0x31f2fd80), DICO(0x3a200480), DICO(0x416bea80), - DICO(0x4ba1ce00), DICO(0x52a88000), DICO(0x58d0db00), DICO(0x5d3e5800), - DICO(0x059e0e70), DICO(0x08166ba0), DICO(0x0c9df590), DICO(0x11de5f40), - DICO(0x14c08ea0), DICO(0x1c5ad6e0), DICO(0x24178ec0), DICO(0x28eb7640), - DICO(0x31626280), DICO(0x35d43100), DICO(0x3cad10c0), DICO(0x422e1480), - DICO(0x49baee00), DICO(0x53dd7180), DICO(0x5a17bb80), DICO(0x5e4bb180), - DICO(0x03c64d00), DICO(0x05de0b28), DICO(0x0ccb82b0), DICO(0x144b1c00), - DICO(0x19e39ec0), DICO(0x21e795c0), DICO(0x28149380), DICO(0x2f1ff7c0), - DICO(0x35b7c900), DICO(0x3c5f4800), DICO(0x42799c00), DICO(0x48746180), - DICO(0x4e8f4d00), DICO(0x53b98380), DICO(0x58d60000), DICO(0x5d71a000), - DICO(0x050a2990), DICO(0x071bdb88), DICO(0x0f0d76b0), DICO(0x17a78de0), - DICO(0x1aa20720), DICO(0x22eea3c0), DICO(0x276e9640), DICO(0x2ecc4d80), - DICO(0x335f7900), DICO(0x37838640), DICO(0x3c81be80), DICO(0x41a4e400), - DICO(0x476fa280), DICO(0x4f2ab780), DICO(0x56e87b80), DICO(0x5cbf2900), - DICO(0x06724c98), DICO(0x099f5e20), DICO(0x119bcec0), DICO(0x16be11a0), - DICO(0x19d53b00), DICO(0x1eb51360), DICO(0x2302c300), DICO(0x29c20d40), - DICO(0x30dd5200), DICO(0x36576a80), DICO(0x3e3e7540), DICO(0x45aa9f80), - DICO(0x4c833300), DICO(0x51d7ed00), DICO(0x57a4a380), DICO(0x5cce9100), - DICO(0x05588b60), DICO(0x075a70a0), DICO(0x0ef96e30), DICO(0x12ac1100), - DICO(0x1649dde0), DICO(0x1a50cdc0), DICO(0x22486140), DICO(0x27c7c800), - DICO(0x2e08bd80), DICO(0x355d20c0), DICO(0x3925a9c0), DICO(0x44e2b480), - DICO(0x4fa4e680), DICO(0x5470e180), DICO(0x595fee80), DICO(0x5d672f00), - DICO(0x04a781c0), DICO(0x060a89f0), DICO(0x111f56a0), DICO(0x16b93be0), - DICO(0x19ce9120), DICO(0x1e1fda60), DICO(0x2259f880), DICO(0x285cad80), - DICO(0x2b140ec0), DICO(0x2f069940), DICO(0x33bbce40), DICO(0x3dd3e6c0), - DICO(0x49154880), DICO(0x5008e380), DICO(0x568c4680), DICO(0x5da43780), - DICO(0x053ade38), DICO(0x0708a200), DICO(0x0bad43d0), DICO(0x1860cf40), - DICO(0x1c40dbe0), DICO(0x1f5a0120), DICO(0x25bcf7c0), DICO(0x29e0c840), - DICO(0x2f223840), DICO(0x390c8940), DICO(0x3e1d7340), DICO(0x41c6a000), - DICO(0x46cc2880), DICO(0x5006e280), DICO(0x5744e180), DICO(0x5d297780), - DICO(0x0401a3f0), DICO(0x07be4a08), DICO(0x14f5f1a0), DICO(0x1c348080), - DICO(0x226753c0), DICO(0x274b1b80), DICO(0x2c11a300), DICO(0x30798c00), - DICO(0x35598c80), DICO(0x3aab18c0), DICO(0x4030f380), DICO(0x45c7b400), - DICO(0x4be5be00), DICO(0x5180db80), DICO(0x57613b00), DICO(0x5cffff80), - DICO(0x084b3090), DICO(0x0fe5b3b0), DICO(0x16dfd480), DICO(0x1d0aa700), - DICO(0x24e08180), DICO(0x2b498640), DICO(0x30885b80), DICO(0x34ccc480), - DICO(0x38fedec0), DICO(0x3cdf9c40), DICO(0x41737600), DICO(0x46ab9800), - DICO(0x4c772e80), DICO(0x51e5b980), DICO(0x57df9900), DICO(0x5d5d7180), - DICO(0x09549f00), DICO(0x12b84da0), DICO(0x1aeaf1c0), DICO(0x2142a9c0), - DICO(0x275c9a00), DICO(0x2c260c80), DICO(0x3038c700), DICO(0x34081d00), - DICO(0x38612f40), DICO(0x3d0bf8c0), DICO(0x42473e00), DICO(0x47ecad80), - DICO(0x4db34380), DICO(0x52f0ab00), DICO(0x584cc980), DICO(0x5d62ae80), - DICO(0x0aeca1e0), DICO(0x14354700), DICO(0x19955ba0), DICO(0x1de331e0), - DICO(0x21cd60c0), DICO(0x25e72d80), DICO(0x2a402880), DICO(0x2efa64c0), - DICO(0x3478d9c0), DICO(0x3a889e80), DICO(0x40744e00), DICO(0x4626fe80), - DICO(0x4bd80900), DICO(0x51120f00), DICO(0x56d8d280), DICO(0x5c654400), - DICO(0x07a40af8), DICO(0x0e65ec20), DICO(0x14c76780), DICO(0x19b35a60), - DICO(0x1f76b7c0), DICO(0x24f512c0), DICO(0x2ae89f00), DICO(0x3036c3c0), - DICO(0x36041800), DICO(0x3bc4df80), DICO(0x41adb080), DICO(0x477fbb80), - DICO(0x4d5aa480), DICO(0x52cb0600), DICO(0x586f7280), DICO(0x5db40180), - DICO(0x03997610), DICO(0x06175db0), DICO(0x0ea8c9c0), DICO(0x14397000), - DICO(0x1ba34f60), DICO(0x22346680), DICO(0x28958080), DICO(0x2ea76840), - DICO(0x33ee68c0), DICO(0x39e769c0), DICO(0x3f862500), DICO(0x4579b080), - DICO(0x4b49cd00), DICO(0x5107da80), DICO(0x573cde00), DICO(0x5d090780), - DICO(0x046fc2f8), DICO(0x0640dbc0), DICO(0x09da7ab0), DICO(0x174e4220), - DICO(0x23dc8cc0), DICO(0x27016200), DICO(0x2a9a5240), DICO(0x2e6cc7c0), - DICO(0x321ced40), DICO(0x38ca2d00), DICO(0x41482680), DICO(0x451e3700), - DICO(0x4b90d800), DICO(0x50d0ba80), DICO(0x55602e80), DICO(0x5a465200), - DICO(0x03ca7e28), DICO(0x057c9dd0), DICO(0x0c9ff0e0), DICO(0x1957fae0), - DICO(0x1fef7860), DICO(0x27920c80), DICO(0x2cb233c0), DICO(0x32015c80), - DICO(0x36af4f40), DICO(0x3ac18240), DICO(0x3f93d8c0), DICO(0x44eaef80), - DICO(0x4aab5d80), DICO(0x50840e80), DICO(0x56c4cc80), DICO(0x5cb26600), - DICO(0x038d53f8), DICO(0x050d1118), DICO(0x0bc14690), DICO(0x18918000), - DICO(0x1e2a6ee0), DICO(0x24cc0c00), DICO(0x2a767d40), DICO(0x2e614940), - DICO(0x32859c40), DICO(0x377fd940), DICO(0x3d3a3e40), DICO(0x43e81380), - DICO(0x4aaac080), DICO(0x509e7800), DICO(0x57023a00), DICO(0x5d733c00), - DICO(0x04cfa5e0), DICO(0x06deeb38), DICO(0x0a501c40), DICO(0x136d8aa0), - DICO(0x17f16e40), DICO(0x1c119300), DICO(0x26154b00), DICO(0x2a0da100), - DICO(0x2f5935c0), DICO(0x37108d40), DICO(0x3aef07c0), DICO(0x3fccf340), - DICO(0x47e4a080), DICO(0x4d8de100), DICO(0x54eb6980), DICO(0x5cdb5380)}; - -/* ACELP: table for decoding - adaptive codebook gain g_p (left column). Scaled by 2.0f. - innovative codebook gain g_c (right column). Scaled by 16.0f. -*/ -const FIXP_SGL fdk_t_qua_gain7b[128 * 2] = { - 204, 441, 464, 1977, 869, 1077, 1072, 3062, 1281, 4759, 1647, - 1539, 1845, 7020, 1853, 634, 1995, 2336, 2351, 15400, 2661, 1165, - 2702, 3900, 2710, 10133, 3195, 1752, 3498, 2624, 3663, 849, 3984, - 5697, 4214, 3399, 4415, 1304, 4695, 2056, 5376, 4558, 5386, 676, - 5518, 23554, 5567, 7794, 5644, 3061, 5672, 1513, 5957, 2338, 6533, - 1060, 6804, 5998, 6820, 1767, 6937, 3837, 7277, 414, 7305, 2665, - 7466, 11304, 7942, 794, 8007, 1982, 8007, 1366, 8326, 3105, 8336, - 4810, 8708, 7954, 8989, 2279, 9031, 1055, 9247, 3568, 9283, 1631, - 9654, 6311, 9811, 2605, 10120, 683, 10143, 4179, 10245, 1946, 10335, - 1218, 10468, 9960, 10651, 3000, 10951, 1530, 10969, 5290, 11203, 2305, - 11325, 3562, 11771, 6754, 11839, 1849, 11941, 4495, 11954, 1298, 11975, - 15223, 11977, 883, 11986, 2842, 12438, 2141, 12593, 3665, 12636, 8367, - 12658, 1594, 12886, 2628, 12984, 4942, 13146, 1115, 13224, 524, 13341, - 3163, 13399, 1923, 13549, 5961, 13606, 1401, 13655, 2399, 13782, 3909, - 13868, 10923, 14226, 1723, 14232, 2939, 14278, 7528, 14439, 4598, 14451, - 984, 14458, 2265, 14792, 1403, 14818, 3445, 14899, 5709, 15017, 15362, - 15048, 1946, 15069, 2655, 15405, 9591, 15405, 4079, 15570, 7183, 15687, - 2286, 15691, 1624, 15699, 3068, 15772, 5149, 15868, 1205, 15970, 696, - 16249, 3584, 16338, 1917, 16424, 2560, 16483, 4438, 16529, 6410, 16620, - 11966, 16839, 8780, 17030, 3050, 17033, 18325, 17092, 1568, 17123, 5197, - 17351, 2113, 17374, 980, 17566, 26214, 17609, 3912, 17639, 32767, 18151, - 7871, 18197, 2516, 18202, 5649, 18679, 3283, 18930, 1370, 19271, 13757, - 19317, 4120, 19460, 1973, 19654, 10018, 19764, 6792, 19912, 5135, 20040, - 2841, 21234, 19833}; - -/* ACELP: factor table for interpolation of LPC coeffs in LSP domain */ -const FIXP_SGL lsp_interpol_factor[2][NB_SUBFR] = { - {FL2FXCONST_SGL(0.125f), FL2FXCONST_SGL(0.375f), FL2FXCONST_SGL(0.625f), - FL2FXCONST_SGL(0.875f)}, /* for coreCoderFrameLength = 1024 */ - {FL2FXCONST_SGL(0.166667f), FL2FXCONST_SGL(0.5f), FL2FXCONST_SGL(0.833333f), - 0x0} /* for coreCoderFrameLength = 768 */ -}; - -/* For bass post filter */ -#ifndef TABLE_filt_lp -const FIXP_SGL fdk_dec_filt_lp[1 + L_FILT] = { - FL2FXCONST_SGL_FILT(0.088250f), FL2FXCONST_SGL_FILT(0.086410f), - FL2FXCONST_SGL_FILT(0.081074f), FL2FXCONST_SGL_FILT(0.072768f), - FL2FXCONST_SGL_FILT(0.062294f), FL2FXCONST_SGL_FILT(0.050623f), - FL2FXCONST_SGL_FILT(0.038774f), FL2FXCONST_SGL_FILT(0.027692f), - FL2FXCONST_SGL_FILT(0.018130f), FL2FXCONST_SGL_FILT(0.010578f), - FL2FXCONST_SGL_FILT(0.005221f), FL2FXCONST_SGL_FILT(0.001946f), - FL2FXCONST_SGL_FILT(0.000385f)}; -#endif - -/* FAC window tables for coreCoderFrameLength = 1024 */ -const FIXP_WTB FacWindowSynth128[] = { - WTC(0x7fff6216), WTC(0x7ffa72d1), WTC(0x7ff09478), WTC(0x7fe1c76b), - WTC(0x7fce0c3e), WTC(0x7fb563b3), WTC(0x7f97cebd), WTC(0x7f754e80), - WTC(0x7f4de451), WTC(0x7f2191b4), WTC(0x7ef05860), WTC(0x7eba3a39), - WTC(0x7e7f3957), WTC(0x7e3f57ff), WTC(0x7dfa98a8), WTC(0x7db0fdf8), - WTC(0x7d628ac6), WTC(0x7d0f4218), WTC(0x7cb72724), WTC(0x7c5a3d50), - WTC(0x7bf88830), WTC(0x7b920b89), WTC(0x7b26cb4f), WTC(0x7ab6cba4), - WTC(0x7a4210d8), WTC(0x79c89f6e), WTC(0x794a7c12), WTC(0x78c7aba2), - WTC(0x78403329), WTC(0x77b417df), WTC(0x77235f2d), WTC(0x768e0ea6), - WTC(0x75f42c0b), WTC(0x7555bd4c), WTC(0x74b2c884), WTC(0x740b53fb), - WTC(0x735f6626), WTC(0x72af05a7), WTC(0x71fa3949), WTC(0x71410805), - WTC(0x708378ff), WTC(0x6fc19385), WTC(0x6efb5f12), WTC(0x6e30e34a), - WTC(0x6d6227fa), WTC(0x6c8f351c), WTC(0x6bb812d1), WTC(0x6adcc964), - WTC(0x69fd614a), WTC(0x6919e320), WTC(0x683257ab), WTC(0x6746c7d8), - WTC(0x66573cbb), WTC(0x6563bf92), WTC(0x646c59bf), WTC(0x637114cc), - WTC(0x6271fa69), WTC(0x616f146c), WTC(0x60686ccf), WTC(0x5f5e0db3), - WTC(0x5e50015d), WTC(0x5d3e5237), WTC(0x5c290acc), WTC(0x5b1035cf), - WTC(0x59f3de12), WTC(0x58d40e8c), WTC(0x57b0d256), WTC(0x568a34a9), - WTC(0x556040e2), WTC(0x5433027d), WTC(0x53028518), WTC(0x51ced46e), - WTC(0x5097fc5e), WTC(0x4f5e08e3), WTC(0x4e210617), WTC(0x4ce10034), - WTC(0x4b9e0390), WTC(0x4a581c9e), WTC(0x490f57ee), WTC(0x47c3c22f), - WTC(0x46756828), WTC(0x452456bd), WTC(0x43d09aed), WTC(0x427a41d0), - WTC(0x4121589b), WTC(0x3fc5ec98), WTC(0x3e680b2c), WTC(0x3d07c1d6), - WTC(0x3ba51e29), WTC(0x3a402dd2), WTC(0x38d8fe93), WTC(0x376f9e46), - WTC(0x36041ad9), WTC(0x34968250), WTC(0x3326e2c3), WTC(0x31b54a5e), - WTC(0x3041c761), WTC(0x2ecc681e), WTC(0x2d553afc), WTC(0x2bdc4e6f), - WTC(0x2a61b101), WTC(0x28e5714b), WTC(0x27679df4), WTC(0x25e845b6), - WTC(0x24677758), WTC(0x22e541af), WTC(0x2161b3a0), WTC(0x1fdcdc1b), - WTC(0x1e56ca1e), WTC(0x1ccf8cb3), WTC(0x1b4732ef), WTC(0x19bdcbf3), - WTC(0x183366e9), WTC(0x16a81305), WTC(0x151bdf86), WTC(0x138edbb1), - WTC(0x120116d5), WTC(0x1072a048), WTC(0x0ee38766), WTC(0x0d53db92), - WTC(0x0bc3ac35), WTC(0x0a3308bd), WTC(0x08a2009a), WTC(0x0710a345), - WTC(0x057f0035), WTC(0x03ed26e6), WTC(0x025b26d7), WTC(0x00c90f88), -}; -const FIXP_WTB FacWindowZir128[] = { - WTC(0x7f36f078), WTC(0x7da4d929), WTC(0x7c12d91a), WTC(0x7a80ffcb), - WTC(0x78ef5cbb), WTC(0x775dff66), WTC(0x75ccf743), WTC(0x743c53cb), - WTC(0x72ac246e), WTC(0x711c789a), WTC(0x6f8d5fb8), WTC(0x6dfee92b), - WTC(0x6c71244f), WTC(0x6ae4207a), WTC(0x6957ecfb), WTC(0x67cc9917), - WTC(0x6642340d), WTC(0x64b8cd11), WTC(0x6330734d), WTC(0x61a935e2), - WTC(0x602323e5), WTC(0x5e9e4c60), WTC(0x5d1abe51), WTC(0x5b9888a8), - WTC(0x5a17ba4a), WTC(0x5898620c), WTC(0x571a8eb5), WTC(0x559e4eff), - WTC(0x5423b191), WTC(0x52aac504), WTC(0x513397e2), WTC(0x4fbe389f), - WTC(0x4e4ab5a2), WTC(0x4cd91d3d), WTC(0x4b697db0), WTC(0x49fbe527), - WTC(0x489061ba), WTC(0x4727016d), WTC(0x45bfd22e), WTC(0x445ae1d7), - WTC(0x42f83e2a), WTC(0x4197f4d4), WTC(0x403a1368), WTC(0x3edea765), - WTC(0x3d85be30), WTC(0x3c2f6513), WTC(0x3adba943), WTC(0x398a97d8), - WTC(0x383c3dd1), WTC(0x36f0a812), WTC(0x35a7e362), WTC(0x3461fc70), - WTC(0x331effcc), WTC(0x31def9e9), WTC(0x30a1f71d), WTC(0x2f6803a2), - WTC(0x2e312b92), WTC(0x2cfd7ae8), WTC(0x2bccfd83), WTC(0x2a9fbf1e), - WTC(0x2975cb57), WTC(0x284f2daa), WTC(0x272bf174), WTC(0x260c21ee), - WTC(0x24efca31), WTC(0x23d6f534), WTC(0x22c1adc9), WTC(0x21affea3), - WTC(0x20a1f24d), WTC(0x1f979331), WTC(0x1e90eb94), WTC(0x1d8e0597), - WTC(0x1c8eeb34), WTC(0x1b93a641), WTC(0x1a9c406e), WTC(0x19a8c345), - WTC(0x18b93828), WTC(0x17cda855), WTC(0x16e61ce0), WTC(0x16029eb6), - WTC(0x1523369c), WTC(0x1447ed2f), WTC(0x1370cae4), WTC(0x129dd806), - WTC(0x11cf1cb6), WTC(0x1104a0ee), WTC(0x103e6c7b), WTC(0x0f7c8701), - WTC(0x0ebef7fb), WTC(0x0e05c6b7), WTC(0x0d50fa59), WTC(0x0ca099da), - WTC(0x0bf4ac05), WTC(0x0b4d377c), WTC(0x0aaa42b4), WTC(0x0a0bd3f5), - WTC(0x0971f15a), WTC(0x08dca0d3), WTC(0x084be821), WTC(0x07bfccd7), - WTC(0x0738545e), WTC(0x06b583ee), WTC(0x06376092), WTC(0x05bdef28), - WTC(0x0549345c), WTC(0x04d934b1), WTC(0x046df477), WTC(0x040777d0), - WTC(0x03a5c2b0), WTC(0x0348d8dc), WTC(0x02f0bde8), WTC(0x029d753a), - WTC(0x024f0208), WTC(0x02056758), WTC(0x01c0a801), WTC(0x0180c6a9), - WTC(0x0145c5c7), WTC(0x010fa7a0), WTC(0x00de6e4c), WTC(0x00b21baf), - WTC(0x008ab180), WTC(0x00683143), WTC(0x004a9c4d), WTC(0x0031f3c2), - WTC(0x001e3895), WTC(0x000f6b88), WTC(0x00058d2f), WTC(0x00009dea), -}; -const FIXP_WTB FacWindowSynth64[] = { - WTC(0x7ffd885a), WTC(0x7fe9cbc0), WTC(0x7fc25596), WTC(0x7f872bf3), - WTC(0x7f3857f6), WTC(0x7ed5e5c6), WTC(0x7e5fe493), WTC(0x7dd6668f), - WTC(0x7d3980ec), WTC(0x7c894bde), WTC(0x7bc5e290), WTC(0x7aef6323), - WTC(0x7a05eead), WTC(0x7909a92d), WTC(0x77fab989), WTC(0x76d94989), - WTC(0x75a585cf), WTC(0x745f9dd1), WTC(0x7307c3d0), WTC(0x719e2cd2), - WTC(0x7023109a), WTC(0x6e96a99d), WTC(0x6cf934fc), WTC(0x6b4af279), - WTC(0x698c246c), WTC(0x67bd0fbd), WTC(0x65ddfbd3), WTC(0x63ef3290), - WTC(0x61f1003f), WTC(0x5fe3b38d), WTC(0x5dc79d7c), WTC(0x5b9d1154), - WTC(0x59646498), WTC(0x571deefa), WTC(0x54ca0a4b), WTC(0x5269126e), - WTC(0x4ffb654d), WTC(0x4d8162c4), WTC(0x4afb6c98), WTC(0x4869e665), - WTC(0x45cd358f), WTC(0x4325c135), WTC(0x4073f21d), WTC(0x3db832a6), - WTC(0x3af2eeb7), WTC(0x382493b0), WTC(0x354d9057), WTC(0x326e54c7), - WTC(0x2f875262), WTC(0x2c98fbba), WTC(0x29a3c485), WTC(0x26a82186), - WTC(0x23a6887f), WTC(0x209f701c), WTC(0x1d934fe5), WTC(0x1a82a026), - WTC(0x176dd9de), WTC(0x145576b1), WTC(0x1139f0cf), WTC(0x0e1bc2e4), - WTC(0x0afb6805), WTC(0x07d95b9e), WTC(0x04b6195d), WTC(0x01921d20), -}; -const FIXP_WTB FacWindowZir64[] = { - WTC(0x7e6de2e0), WTC(0x7b49e6a3), WTC(0x7826a462), WTC(0x750497fb), - WTC(0x71e43d1c), WTC(0x6ec60f31), WTC(0x6baa894f), WTC(0x68922622), - WTC(0x657d5fda), WTC(0x626cb01b), WTC(0x5f608fe4), WTC(0x5c597781), - WTC(0x5957de7a), WTC(0x565c3b7b), WTC(0x53670446), WTC(0x5078ad9e), - WTC(0x4d91ab39), WTC(0x4ab26fa9), WTC(0x47db6c50), WTC(0x450d1149), - WTC(0x4247cd5a), WTC(0x3f8c0de3), WTC(0x3cda3ecb), WTC(0x3a32ca71), - WTC(0x3796199b), WTC(0x35049368), WTC(0x327e9d3c), WTC(0x30049ab3), - WTC(0x2d96ed92), WTC(0x2b35f5b5), WTC(0x28e21106), WTC(0x269b9b68), - WTC(0x2462eeac), WTC(0x22386284), WTC(0x201c4c73), WTC(0x1e0effc1), - WTC(0x1c10cd70), WTC(0x1a22042d), WTC(0x1842f043), WTC(0x1673db94), - WTC(0x14b50d87), WTC(0x1306cb04), WTC(0x11695663), WTC(0x0fdcef66), - WTC(0x0e61d32e), WTC(0x0cf83c30), WTC(0x0ba0622f), WTC(0x0a5a7a31), - WTC(0x0926b677), WTC(0x08054677), WTC(0x06f656d3), WTC(0x05fa1153), - WTC(0x05109cdd), WTC(0x043a1d70), WTC(0x0376b422), WTC(0x02c67f14), - WTC(0x02299971), WTC(0x01a01b6d), WTC(0x012a1a3a), WTC(0x00c7a80a), - WTC(0x0078d40d), WTC(0x003daa6a), WTC(0x00163440), WTC(0x000277a6), -}; -const FIXP_WTB FacWindowSynth32[] = { - WTC(0x7ff62182), WTC(0x7fa736b4), WTC(0x7f0991c4), WTC(0x7e1d93ea), - WTC(0x7ce3ceb2), WTC(0x7b5d039e), WTC(0x798a23b1), WTC(0x776c4edb), - WTC(0x7504d345), WTC(0x72552c85), WTC(0x6f5f02b2), WTC(0x6c242960), - WTC(0x68a69e81), WTC(0x64e88926), WTC(0x60ec3830), WTC(0x5cb420e0), - WTC(0x5842dd54), WTC(0x539b2af0), WTC(0x4ebfe8a5), WTC(0x49b41533), - WTC(0x447acd50), WTC(0x3f1749b8), WTC(0x398cdd32), WTC(0x33def287), - WTC(0x2e110a62), WTC(0x2826b928), WTC(0x2223a4c5), WTC(0x1c0b826a), - WTC(0x15e21445), WTC(0x0fab272b), WTC(0x096a9049), WTC(0x03242abf), -}; -const FIXP_WTB FacWindowZir32[] = { - WTC(0x7cdbd541), WTC(0x76956fb7), WTC(0x7054d8d5), WTC(0x6a1debbb), - WTC(0x63f47d96), WTC(0x5ddc5b3b), WTC(0x57d946d8), WTC(0x51eef59e), - WTC(0x4c210d79), WTC(0x467322ce), WTC(0x40e8b648), WTC(0x3b8532b0), - WTC(0x364beacd), WTC(0x3140175b), WTC(0x2c64d510), WTC(0x27bd22ac), - WTC(0x234bdf20), WTC(0x1f13c7d0), WTC(0x1b1776da), WTC(0x1759617f), - WTC(0x13dbd6a0), WTC(0x10a0fd4e), WTC(0x0daad37b), WTC(0x0afb2cbb), - WTC(0x0893b125), WTC(0x0675dc4f), WTC(0x04a2fc62), WTC(0x031c314e), - WTC(0x01e26c16), WTC(0x00f66e3c), WTC(0x0058c94c), WTC(0x0009de7e), -}; - -/* FAC window tables for coreCoderFrameLength = 768 */ -const FIXP_WTB FacWindowSynth96[] = { - WTC(0x7ffee744), WTC(0x7ff62182), WTC(0x7fe49698), WTC(0x7fca47b9), - WTC(0x7fa736b4), WTC(0x7f7b65ef), WTC(0x7f46d86c), WTC(0x7f0991c4), - WTC(0x7ec3962a), WTC(0x7e74ea6a), WTC(0x7e1d93ea), WTC(0x7dbd98a4), - WTC(0x7d54ff2e), WTC(0x7ce3ceb2), WTC(0x7c6a0ef2), WTC(0x7be7c847), - WTC(0x7b5d039e), WTC(0x7ac9ca7a), WTC(0x7a2e26f2), WTC(0x798a23b1), - WTC(0x78ddcbf5), WTC(0x78292b8d), WTC(0x776c4edb), WTC(0x76a742d1), - WTC(0x75da14ef), WTC(0x7504d345), WTC(0x74278c72), WTC(0x73424fa0), - WTC(0x72552c85), WTC(0x71603361), WTC(0x706374ff), WTC(0x6f5f02b2), - WTC(0x6e52ee52), WTC(0x6d3f4a40), WTC(0x6c242960), WTC(0x6b019f1a), - WTC(0x69d7bf57), WTC(0x68a69e81), WTC(0x676e5183), WTC(0x662eedc3), - WTC(0x64e88926), WTC(0x639b3a0b), WTC(0x62471749), WTC(0x60ec3830), - WTC(0x5f8ab487), WTC(0x5e22a487), WTC(0x5cb420e0), WTC(0x5b3f42ae), - WTC(0x59c42381), WTC(0x5842dd54), WTC(0x56bb8a90), WTC(0x552e4605), - WTC(0x539b2af0), WTC(0x520254ef), WTC(0x5063e008), WTC(0x4ebfe8a5), - WTC(0x4d168b8b), WTC(0x4b67e5e4), WTC(0x49b41533), WTC(0x47fb3757), - WTC(0x463d6a87), WTC(0x447acd50), WTC(0x42b37e96), WTC(0x40e79d8c), - WTC(0x3f1749b8), WTC(0x3d42a2ec), WTC(0x3b69c947), WTC(0x398cdd32), - WTC(0x37abff5d), WTC(0x35c750bc), WTC(0x33def287), WTC(0x31f30638), - WTC(0x3003ad85), WTC(0x2e110a62), WTC(0x2c1b3efb), WTC(0x2a226db5), - WTC(0x2826b928), WTC(0x26284422), WTC(0x2427319d), WTC(0x2223a4c5), - WTC(0x201dc0ef), WTC(0x1e15a99a), WTC(0x1c0b826a), WTC(0x19ff6f2a), - WTC(0x17f193c5), WTC(0x15e21445), WTC(0x13d114d0), WTC(0x11beb9aa), - WTC(0x0fab272b), WTC(0x0d9681c2), WTC(0x0b80edf1), WTC(0x096a9049), - WTC(0x07538d6b), WTC(0x053c0a01), WTC(0x03242abf), WTC(0x010c1460), -}; -const FIXP_WTB FacWindowZir96[] = { - WTC(0x7ef3eba0), WTC(0x7cdbd541), WTC(0x7ac3f5ff), WTC(0x78ac7295), - WTC(0x76956fb7), WTC(0x747f120f), WTC(0x72697e3e), WTC(0x7054d8d5), - WTC(0x6e414656), WTC(0x6c2eeb30), WTC(0x6a1debbb), WTC(0x680e6c3b), - WTC(0x660090d6), WTC(0x63f47d96), WTC(0x61ea5666), WTC(0x5fe23f11), - WTC(0x5ddc5b3b), WTC(0x5bd8ce63), WTC(0x59d7bbde), WTC(0x57d946d8), - WTC(0x55dd924b), WTC(0x53e4c105), WTC(0x51eef59e), WTC(0x4ffc527b), - WTC(0x4e0cf9c8), WTC(0x4c210d79), WTC(0x4a38af44), WTC(0x485400a3), - WTC(0x467322ce), WTC(0x449636b9), WTC(0x42bd5d14), WTC(0x40e8b648), - WTC(0x3f186274), WTC(0x3d4c816a), WTC(0x3b8532b0), WTC(0x39c29579), - WTC(0x3804c8a9), WTC(0x364beacd), WTC(0x34981a1c), WTC(0x32e97475), - WTC(0x3140175b), WTC(0x2f9c1ff8), WTC(0x2dfdab11), WTC(0x2c64d510), - WTC(0x2ad1b9fb), WTC(0x29447570), WTC(0x27bd22ac), WTC(0x263bdc7f), - WTC(0x24c0bd52), WTC(0x234bdf20), WTC(0x21dd5b79), WTC(0x20754b79), - WTC(0x1f13c7d0), WTC(0x1db8e8b7), WTC(0x1c64c5f5), WTC(0x1b1776da), - WTC(0x19d1123d), WTC(0x1891ae7d), WTC(0x1759617f), WTC(0x162840a9), - WTC(0x14fe60e6), WTC(0x13dbd6a0), WTC(0x12c0b5c0), WTC(0x11ad11ae), - WTC(0x10a0fd4e), WTC(0x0f9c8b01), WTC(0x0e9fcc9f), WTC(0x0daad37b), - WTC(0x0cbdb060), WTC(0x0bd8738e), WTC(0x0afb2cbb), WTC(0x0a25eb11), - WTC(0x0958bd2f), WTC(0x0893b125), WTC(0x07d6d473), WTC(0x0722340b), - WTC(0x0675dc4f), WTC(0x05d1d90e), WTC(0x05363586), WTC(0x04a2fc62), - WTC(0x041837b9), WTC(0x0395f10e), WTC(0x031c314e), WTC(0x02ab00d2), - WTC(0x0242675c), WTC(0x01e26c16), WTC(0x018b1596), WTC(0x013c69d6), - WTC(0x00f66e3c), WTC(0x00b92794), WTC(0x00849a11), WTC(0x0058c94c), - WTC(0x0035b847), WTC(0x001b6968), WTC(0x0009de7e), WTC(0x000118bc), -}; -const FIXP_WTB FacWindowSynth48[] = { - WTC(0x7ffb9d15), WTC(0x7fd8878e), WTC(0x7f92661d), WTC(0x7f294bfd), - WTC(0x7e9d55fc), WTC(0x7deeaa7a), WTC(0x7d1d7958), WTC(0x7c29fbee), - WTC(0x7b1474fd), WTC(0x79dd3098), WTC(0x78848414), WTC(0x770acdec), - WTC(0x757075ac), WTC(0x73b5ebd1), WTC(0x71dba9ab), WTC(0x6fe2313c), - WTC(0x6dca0d14), WTC(0x6b93d02e), WTC(0x694015c3), WTC(0x66cf8120), - WTC(0x6442bd7e), WTC(0x619a7dce), WTC(0x5ed77c8a), WTC(0x5bfa7b82), - WTC(0x590443a7), WTC(0x55f5a4d2), WTC(0x52cf758f), WTC(0x4f9292dc), - WTC(0x4c3fdff4), WTC(0x48d84609), WTC(0x455cb40c), WTC(0x41ce1e65), - WTC(0x3e2d7eb1), WTC(0x3a7bd382), WTC(0x36ba2014), WTC(0x32e96c09), - WTC(0x2f0ac320), WTC(0x2b1f34eb), WTC(0x2727d486), WTC(0x2325b847), - WTC(0x1f19f97b), WTC(0x1b05b40f), WTC(0x16ea0646), WTC(0x12c8106f), - WTC(0x0ea0f48c), WTC(0x0a75d60e), WTC(0x0647d97c), WTC(0x02182427), -}; -const FIXP_WTB FacWindowZir48[] = { - WTC(0x7de7dbd9), WTC(0x79b82684), WTC(0x758a29f2), WTC(0x715f0b74), - WTC(0x6d37ef91), WTC(0x6915f9ba), WTC(0x64fa4bf1), WTC(0x60e60685), - WTC(0x5cda47b9), WTC(0x58d82b7a), WTC(0x54e0cb15), WTC(0x50f53ce0), - WTC(0x4d1693f7), WTC(0x4945dfec), WTC(0x45842c7e), WTC(0x41d2814f), - WTC(0x3e31e19b), WTC(0x3aa34bf4), WTC(0x3727b9f7), WTC(0x33c0200c), - WTC(0x306d6d24), WTC(0x2d308a71), WTC(0x2a0a5b2e), WTC(0x26fbbc59), - WTC(0x2405847e), WTC(0x21288376), WTC(0x1e658232), WTC(0x1bbd4282), - WTC(0x19307ee0), WTC(0x16bfea3d), WTC(0x146c2fd2), WTC(0x1235f2ec), - WTC(0x101dcec4), WTC(0x0e245655), WTC(0x0c4a142f), WTC(0x0a8f8a54), - WTC(0x08f53214), WTC(0x077b7bec), WTC(0x0622cf68), WTC(0x04eb8b03), - WTC(0x03d60412), WTC(0x02e286a8), WTC(0x02115586), WTC(0x0162aa04), - WTC(0x00d6b403), WTC(0x006d99e3), WTC(0x00277872), WTC(0x000462eb), -}; diff --git a/libAACdec/src/usacdec_rom.h b/libAACdec/src/usacdec_rom.h deleted file mode 100644 index f969e90..0000000 --- a/libAACdec/src/usacdec_rom.h +++ /dev/null @@ -1,154 +0,0 @@ -/* ----------------------------------------------------------------------------- -Software License for The Fraunhofer FDK AAC Codec Library for Android - -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten -Forschung e.V. All rights reserved. - - 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software -that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding -scheme for digital audio. This FDK AAC Codec software is intended to be used on -a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient -general perceptual audio codecs. AAC-ELD is considered the best-performing -full-bandwidth communications codec by independent studies and is widely -deployed. AAC has been standardized by ISO and IEC as part of the MPEG -specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including -those of Fraunhofer) may be obtained through Via Licensing -(www.vialicensing.com) or through the respective patent owners individually for -the purpose of encoding or decoding bit streams in products that are compliant -with the ISO/IEC MPEG audio standards. Please note that most manufacturers of -Android devices already license these patent claims through Via Licensing or -directly from the patent owners, and therefore FDK AAC Codec software may -already be covered under those patent licenses when it is used for those -licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions -with enhanced sound quality, are also available from Fraunhofer. Users are -encouraged to check the Fraunhofer website for additional applications -information and documentation. - -2. COPYRIGHT LICENSE - -Redistribution and use in source and binary forms, with or without modification, -are permitted without payment of copyright license fees provided that you -satisfy the following conditions: - -You must retain the complete text of this software license in redistributions of -the FDK AAC Codec or your modifications thereto in source code form. - -You must retain the complete text of this software license in the documentation -and/or other materials provided with redistributions of the FDK AAC Codec or -your modifications thereto in binary form. You must make available free of -charge copies of the complete source code of the FDK AAC Codec and your -modifications thereto to recipients of copies in binary form. - -The name of Fraunhofer may not be used to endorse or promote products derived -from this library without prior written permission. - -You may not charge copyright license fees for anyone to use, copy or distribute -the FDK AAC Codec software or your modifications thereto. - -Your modified versions of the FDK AAC Codec must carry prominent notices stating -that you changed the software and the date of any change. For modified versions -of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" -must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK -AAC Codec Library for Android." - -3. NO PATENT LICENSE - -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without -limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. -Fraunhofer provides no warranty of patent non-infringement with respect to this -software. - -You may use this FDK AAC Codec software or modifications thereto only for -purposes that are authorized by appropriate patent licenses. - -4. DISCLAIMER - -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright -holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, -including but not limited to the implied warranties of merchantability and -fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, -or consequential damages, including but not limited to procurement of substitute -goods or services; loss of use, data, or profits, or business interruption, -however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of -this software, even if advised of the possibility of such damage. - -5. CONTACT INFORMATION - -Fraunhofer Institute for Integrated Circuits IIS -Attention: Audio and Multimedia Departments - FDK AAC LL -Am Wolfsmantel 33 -91058 Erlangen, Germany - -www.iis.fraunhofer.de/amm -amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------ */ - -/**************************** AAC decoder library ****************************** - - Author(s): M. Jander - - Description: re8.h - -*******************************************************************************/ - -#ifndef USACDEC_ROM_H -#define USACDEC_ROM_H - -#include "common_fix.h" -#include "FDK_lpc.h" - -#include "usacdec_const.h" - -/* RE8 lattice quantiser constants */ -#define NB_SPHERE 32 -#define NB_LEADER 37 -#define NB_LDSIGN 226 -#define NB_LDQ3 9 -#define NB_LDQ4 28 - -#define LSF_SCALE 13 - -/* RE8 lattice quantiser tables */ -extern const UINT fdk_dec_tab_factorial[8]; -extern const UCHAR fdk_dec_Ia[NB_LEADER]; -extern const UCHAR fdk_dec_Ds[NB_LDSIGN]; -extern const USHORT fdk_dec_Is[NB_LDSIGN]; -extern const UCHAR fdk_dec_Ns[], fdk_dec_A3[], fdk_dec_A4[]; -extern const UCHAR fdk_dec_Da[][8]; -extern const USHORT fdk_dec_I3[], fdk_dec_I4[]; - -/* temp float tables for LPC decoding */ -extern const FIXP_LPC fdk_dec_lsf_init[16]; -extern const FIXP_LPC fdk_dec_dico_lsf_abs_8b[16 * 256]; - -/* ACELP tables */ -#define SF_QUA_GAIN7B 4 -extern const FIXP_SGL fdk_t_qua_gain7b[128 * 2]; -extern const FIXP_SGL lsp_interpol_factor[2][NB_SUBFR]; - -/* For bass post filter */ -#define L_FILT 12 /* Delay of up-sampling filter */ - -extern const FIXP_SGL fdk_dec_filt_lp[1 + L_FILT]; - -extern const FIXP_WTB FacWindowSynth128[128]; -extern const FIXP_WTB FacWindowZir128[128]; -extern const FIXP_WTB FacWindowSynth64[64]; -extern const FIXP_WTB FacWindowZir64[64]; -extern const FIXP_WTB FacWindowSynth32[32]; -extern const FIXP_WTB FacWindowZir32[32]; -extern const FIXP_WTB FacWindowSynth96[96]; -extern const FIXP_WTB FacWindowZir96[96]; -extern const FIXP_WTB FacWindowSynth48[48]; -extern const FIXP_WTB FacWindowZir48[48]; - -#endif /* USACDEC_ROM_H */ -- cgit v1.1